Skip to content


Switch branches/tags

Name already in use

A tag already exists with the provided branch name. Many Git commands accept both tag and branch names, so creating this branch may cause unexpected behavior. Are you sure you want to create this branch?

Latest commit

In some cases it's possible that some packets still arrive in userspace
immediately after a stream has been pushed to the kernel, for example if
some packets are already in the queue or if there is some processing
delay (e.g. writing to Redis). Allow for a short delay before counting a
stream as userspace if it has been pushed to the kernel.

Change-Id: I55a6e255868c8c2a9e93355a4aa2287f07b3748d

Git stats


Failed to load latest commit information.
Latest commit message
Commit time

Code Testing Debian Package CI Coverity

What is rtpengine?

The Sipwise NGCP rtpengine is a proxy for RTP traffic and other UDP based media traffic. It's meant to be used with the Kamailio SIP proxy and forms a drop-in replacement for any of the other available RTP and media proxies.

Currently the only supported platform is GNU/Linux.

Mailing List

For general questions, discussion, requests for support, and community chat, join our mailing list. Please do not use the Github issue tracker for this purpose.


  • Media traffic running over either IPv4 or IPv6
  • Bridging between IPv4 and IPv6 user agents
  • Bridging between different IP networks or interfaces
  • TOS/QoS field setting
  • Customizable port range
  • Multi-threaded
  • Advertising different addresses for operation behind NAT
  • In-kernel packet forwarding for low-latency and low-CPU performance
  • Automatic fallback to normal userspace operation if kernel module is unavailable
  • Support for Kamailio's rtpproxy module
  • Legacy support for old OpenSER mediaproxy module
  • HTTP, HTTPS, and WebSocket (WS and WSS) interfaces

When used through the rtpengine module (or its older counterpart called rtpproxy-ng), the following additional features are available:

  • Full SDP parsing and rewriting
  • Supports non-standard RTCP ports (RFC 3605)
  • ICE (RFC 5245) support:
    • Bridging between ICE-enabled and ICE-unaware user agents
    • Optionally acting only as additional ICE relay/candidate
    • Optionally forcing relay of media streams by removing other ICE candidates
    • Optionally act as an "ICE lite" peer only
  • SRTP (RFC 3711) support:
    • Support for SDES (RFC 4568) and DTLS-SRTP (RFC 5764)
    • AES-CM and AES-F8 ciphers, both in userspace and in kernel
    • HMAC-SHA1 packet authentication
    • Bridging between RTP and SRTP user agents
    • Opportunistic SRTP (RFC 8643)
    • AES-GCM Authenticated Encryption (AEAD) (RFC 7714)
  • Support for RTCP profile with feedback extensions (RTP/AVPF, RFC 4585 and 5124)
  • Arbitrary bridging between any of the supported RTP profiles (RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF)
  • RTP/RTCP multiplexing (RFC 5761) and demultiplexing
  • Breaking of BUNDLE'd media streams (draft-ietf-mmusic-sdp-bundle-negotiation)
  • Recording of media streams, decrypted if possible
  • Transcoding and repacketization
  • Transcoding between RFC 2833/4733 DTMF event packets and in-band DTMF tones (and vice versa)
  • Injection of DTMF events or PCM DTMF tones into running audio streams
  • Playback of pre-recorded streams/announcements
  • Transcoding between T.38 and PCM (G.711 or other audio codecs)
  • Silence detection and comfort noise (RFC 3389) payloads
  • Media forking
  • Publish/subscribe mechanism for N-to-N media forwarding

There is also limited support for rtpengine to be used as a drop-in replacement for Janus using the native Janus control protocol (see below).

Rtpengine does not (yet) support:

  • ZRTP, although ZRTP passes through rtpengine just fine

Compiling and Installing

Package Repositories

Prebuilt packages for some newer releases of Debian are available on this repository

Compiling on a Debian System

On a Debian system, everything can be built and packaged into Debian packages by executing dpkg-buildpackage (which can be found in the dpkg-dev package) in the main directory. This script will issue an error and stop if any of the dependency packages are not installed. The script dpkg-checkbuilddeps can be used to check missing dependencies. (See the note about G.729 at the end of this section.)

This will produce a number of .deb files, which can then be installed using the dpkg -i command.

The generated files are (with version being built on an amd64 system):

  • ngcp-rtpengine_6.2.0.0+0~mr6.2.0.0_all.deb

    This is a meta-package, which doesn't contain or install anything on its own, but rather only depends on the other packages to be installed. Not strictly necessary to be installed.

  • ngcp-rtpengine-daemon_6.2.0.0+0~mr6.2.0.0_amd64.deb

    This installed the userspace daemon, which is the main workhorse of rtpengine. This is the minimum requirement for anything to work.

  • ngcp-rtpengine-iptables_6.2.0.0+0~mr6.2.0.0_amd64.deb

    Installs the plugin for iptables and ip6tables. Necessary for in-kernel operation.

  • ngcp-rtpengine-kernel-dkms_6.2.0.0+0~mr6.2.0.0_all.deb

    Kernel module, DKMS version of the package. Recommended for in-kernel operation. The kernel module will be compiled against the currently running kernel using DKMS.

  • ngcp-rtpengine-kernel-source_6.2.0.0+0~mr6.2.0.0_all.deb

    If DKMS is unavailable or not desired, then this package will install the sources for the kernel module for manual compilation. Required for in-kernel operation, but only if the DKMS package can't be used.

  • ngcp-rtpengine-recording-daemon_6.2.0.0+0~mr6.2.0.0_amd64.deb

    Optional separate userspace daemon used for call recording features.

  • -dbg... or -dbgsym... packages

    Debugging symbols for the various components. Optional.

For transcoding purposes, Debian provides an additional package libavcodec-extra to replace the regular libavcodec package. It is recommended to install this extra package to offer support for additional codecs.

To support the G.729 codec for transcoding purposes, the external library bcg729 is required. Please see the section on G.729 support below for details.

Manual Compilation

There are 3 main parts to rtpengine plus one optional component, which can be found in the respective subdirectories. Running make on the top source directory will build all parts. Running make check additionally will run the test suite.

  • daemon

    The userspace daemon and workhorse, minimum requirement for anything to work. Running make will compile the binary, which will be called rtpengine. The following software packages including their development headers are required to compile the daemon:

    • pkg-config
    • GLib including GThread and GLib-JSON version 2.x
    • zlib
    • OpenSSL
    • PCRE library
    • XMLRPC-C version 1.16.08 or higher
    • hiredis library
    • gperf
    • libcurl version 3.x or 4.x
    • libevent version 2.x
    • libpcap
    • libsystemd
    • spandsp
    • MySQL or MariaDB client library (optional for media playback and call recording daemon)
    • libiptc library for iptables management (optional)
    • ffmpeg codec libraries for transcoding (optional) such as libavcodec, libavfilter, libswresample
    • bcg729 for full G.729 transcoding support (optional)
    • libmosquitto
    • libwebsockets

    The Makefile contains a few Debian-specific flags, which may have to removed for compilation to be successful. This will not affect operation in any way.

    If you do not wish to (or cannot) compile the optional iptables management feature, the Makefile also contains a switch to disable it. See the --iptables-chain option for a description. The name of the make switch and its default value is with_iptables_option=yes.

    Similarly, the transcoding feature can be excluded via a switch in the Makefile, making it unnecessary to have the ffmpeg libraries installed. The name of the make switch and its default value is with_transcoding=yes.

    Both Makefile switches can be provided to the make system via environment variables, for example by building with the shell command with_transcoding=no make.

  • iptables-extension

    Required for in-kernel packet forwarding.

    With the iptables development headers installed, issuing make will compile the plugin for iptables and ip6tables. The file will be called and needs to be copied into the xtables module directory. The location of this directory can be determined through pkg-config xtables --variable=xtlibdir on newer systems, and/or is usually either /lib/xtables/ or /usr/lib/x86_64-linux-gnu/xtables/.

  • kernel-module

    Required for in-kernel packet forwarding.

    Compilation of the kernel module requires the kernel development headers to be installed in /lib/modules/$VERSION/build/, where $VERSION is the output of the command uname -r. For example, if the command uname -r produces the output 3.9-1-amd64, then the kernel headers must be present in /lib/modules/3.9-1-amd64/build/. The last component of this path (build) is usually a symlink somewhere into /usr/src/, which is fine.

    Successful compilation of the module will produce the file xt_RTPENGINE.ko. The module can be inserted into the running kernel manually through insmod xt_RTPENGINE.ko (which will result in an error if depending modules aren't loaded, for example the x_tables module), but it's recommended to copy the module into /lib/modules/$VERSION/updates/, followed by running depmod -a. After this, the module can be loaded by issuing modprobe xt_RTPENGINE.

  • recording-daemon

    Optional component for the call recording feature. Prerequisites are usage of the kernel module and availability of transcoding (via ffmpeg)


Userspace Daemon

The options are described in detail in the rtpengine(1) man page. If you're reading this on Github, you can view the current master's man page here.

In-kernel Packet Forwarding

In normal userspace-only operation, the overhead involved in processing each individual RTP or media packet is quite significant. This comes from the fact that each time a packet is received on a network interface, the packet must first traverse the stack of the kernel's network protocols, down to locating a process's file descriptor. At this point the linked user process (the daemon) has to be signalled that a new packet is available to be read, the process has to be scheduled to run, once running the process must read the packet, which means it must be copied from kernel space to user space, involving an expensive context switch. Once the packet has been processed by the daemon, it must be sent out again, reversing the whole process.

All this wouldn't be a big deal if it wasn't for the fact that RTP traffic generally consists of many small packets being transferred at high rates. Since the forwarding overhead is incurred on a per-packet basis, the ratio of useful data processed to overhead drops dramatically.

For these reasons, rtpengine provides a kernel module to offload the bulk of the packet forwarding duties from user space to kernel space. Using this technique, a large percentage of the overhead can be eliminated, CPU usage greatly reduced and the number of concurrent calls possible to be handled increased.

In-kernel packet forwarding is implemented as an iptables module (or more precisely, an x_tables module). As such, it comes in two parts, both of which are required for proper operation. One part is the actual kernel module called xt_RTPENGINE. The second part is a plugin to the iptables and ip6tables command-line utilities to make it possible to actually add the required rule to the tables.


In short, the prerequisites for in-kernel packet forwarding are:

  1. The xt_RTPENGINE kernel module must be loaded.
  2. An iptables and/or ip6tables rule must be present in the INPUT chain (or in a custom user-defined chain which is then called by the INPUT chain) to send packets to the RTPENGINE target. This rule should be limited to UDP packets, but otherwise there are no restrictions.
  3. The rtpengine daemon must be running.
  4. All of the above must be set up with the same forwarding table ID (see below).

The sequence of events for a newly established media stream is then:

  1. The SIP proxy (e.g. Kamailio) controls rtpengine and informs it about a newly established call.
  2. The rtpengine daemon allocates local UDP ports and sets up preliminary forward rules based on the info received from the SIP proxy. Only userspace forwarding is set up, nothing is pushed to the kernel module yet.
  3. An RTP packet is received on the local port.
  4. It traverses the iptables chains and gets passed to the xt_RTPENGINE module.
  5. The module doesn't recognize it as belonging to an established stream and thus ignores it.
  6. The packet continues normal processing and eventually ends up in the daemon's receive queue.
  7. The daemon reads it, processes it and forwards it. It also updates some internal data.
  8. This userspace-only processing and forwarding continues for a little while, during which time information about additional streams and/or endpoints may be obtained from the SIP proxy.
  9. After a few seconds, when the daemon is satisfied with what it has learned about the media endpoints, it pushes the forwarding rules to the kernel.
  10. From this moment on, the kernel module will recognize incoming packets belonging to those streams and will forward them on its own. It will stop those packets from traversing the network stacks any further, so the daemon will not see them any more on its receive queues.
  11. In-kernel forwarding is allowed to cease to work at any given time, either accidentally (e.g. by removal of the iptables rule) or deliberately (the daemon will do so in case of a re-invite), in which case forwarding falls back to userspace-only operation.

The Kernel Module

The kernel module supports multiple forwarding tables (not to be confused with the tables managed by iptables), which are identified through their ID number. By default, up to 64 forwarding tables can be created and used, giving them the ID numbers 0 through 63.

Each forwarding table can be thought of a separate proxy instance. Each running instance of the rtpengine daemon controls one such table, and each table can only be controlled by one running instance of the daemon at any given time. In the most common setup, there will be only a single instance of the daemon running and there will be only a single forwarding table in use, with ID zero.

The kernel module can be loaded with the command modprobe xt_RTPENGINE. With the module loaded, a new directory will appear in /proc/, namely /proc/rtpengine/. After loading, the directory will contain only two pseudo-files, control and list. The control file is write-only and is used to create and delete forwarding tables, while the list file is read-only and will produce a list of currently active forwarding tables. With no tables active, it will produce an empty output.

The control pseudo-file supports two commands, add and del, each followed by the forwarding table ID number. To manually create a forwarding table with ID 42, the following command can be used:

echo 'add 42' > /proc/rtpengine/control

After this, the list pseudo-file will produce the single line 42 as output. This will also create a directory called 42 in /proc/rtpengine/, which contains additional pseudo-files to control this particular forwarding table.

To delete this forwarding table, the command del 42 can be issued like above. This will only work if no rtpengine daemon is currently running and controlling this table.

Each subdirectory /proc/rtpengine/$ID/ corresponding to each forwarding table contains the pseudo-files blist, control, list and status. The control file is write-only while the others are read-only. The control file will be kept open by the rtpengine daemon while it's running to issue updates to the forwarding rules during runtime. The daemon also reads the blist file on a regular basis, which produces a list of currently active forwarding rules together with their stats and other details within that table in a binary format. The same output, but in human-readable format, can be obtained by reading the list file. Lastly, the status file produces a short stats output for the forwarding table.

Manual creation of forwarding tables is normally not required as the daemon will do so itself, however deletion of tables may be required after shutdown of the daemon or before a restart to ensure that the daemon can create the table it wants to use.

The kernel module can be unloaded through rmmod xt_RTPENGINE, however this only works if no forwarding table currently exists and no iptables rule currently exists.

The iptables module

In order for the kernel module to be able to actually forward packets, an iptables rule must be set up to send packets into the module. Each such rule is associated with one forwarding table. In the simplest case, for forwarding table 42, this can be done through:

iptables -I INPUT -p udp -j RTPENGINE --id 42

If IPv6 traffic is expected, the same should be done using ip6tables.

It is possible but not strictly necessary to restrict the rules to the UDP port range used by rtpengine, e.g. by supplying a parameter like --dport 30000:40000. If the kernel module receives a packet that it doesn't recognize as belonging to an active media stream, it will simply ignore it and hand it back to the network stack for normal processing.

The RTPENGINE rule need not necessarily be present directly in the INPUT chain. It can also be in a user-defined chain which is then referenced by the INPUT chain, like so:

iptables -N rtpengine
iptables -I INPUT -p udp -j rtpengine
iptables -I rtpengine -j RTPENGINE --id 42

This can be a useful setup if certain firewall scripts are being used.


A typical start-up sequence including in-kernel forwarding might look like this:

# this only needs to be one once after system (re-) boot
modprobe xt_RTPENGINE
iptables -I INPUT -p udp -j RTPENGINE --id 0
ip6tables -I INPUT -p udp -j RTPENGINE --id 0

# ensure that the table we want to use doesn't exist - usually needed after a daemon
# restart, otherwise will error
echo 'del 0' > /proc/rtpengine/control

# start daemon
/usr/bin/rtpengine --table=0 --interface= --interface=2001:db8::4f3:3d \
--listen-ng= --tos=184 --pidfile=/run/ --no-fallback

Running Multiple Instances

In some cases it may be desired to run multiple instances of rtpengine on the same machine, for example if the host is multi-homed and has multiple usable network interfaces with different addresses. This is supported by running multiple instances of the daemon using different command-line options (different local addresses and different listening ports), together with multiple different kernel forwarding tables.

For example, if one local network interface has address and another has address, then the start-up sequence might look like this:

modprobe xt_RTPENGINE
iptables -I INPUT -p udp -d -j RTPENGINE --id 0
iptables -I INPUT -p udp -d -j RTPENGINE --id 1

echo 'del 0' > /proc/rtpengine/control
echo 'del 1' > /proc/rtpengine/control

/usr/bin/rtpengine --table=0 --interface= \
--listen-ng= --tos=184 --pidfile=/run/ --no-fallback
/usr/bin/rtpengine --table=1 --interface= \
--listen-ng= --tos=184 --pidfile=/run/ --no-fallback

With this setup, the SIP proxy can choose which instance of rtpengine to talk to and thus which local interface to use by sending its control messages to either port 2223 or port 2224.


Currently transcoding is supported for audio streams. The feature can be disabled on a compile-time basis, and is enabled by default.

Even though the transcoding feature is available by default, it is not automatically engaged for normal calls. Normally rtpengine leaves codec negotiation up to the clients involved in the call and does not interfere. In this case, if the clients fail to agree on a codec, the call will fail.

The transcoding feature can be engaged for a call by instructing rtpengine to do so by using one of the transcoding options in the ng control protocol, such as transcode or ptime (see below). If a codec is requested via the transcode option that was not originally offered, transcoding will be engaged for that call.

With transcoding active for a call, all unsupported codecs will be removed from the SDP. Transcoding happens in userspace only, so in-kernel packet forwarding will not be available for transcoded codecs. However, even if the transcoding feature has been engaged for a call, not all codecs will necessarily end up being transcoded. Codecs that are supported by both sides will simply be passed through transparently (unless repacketization is active). In-kernel packet forwarding will still be available for these codecs.

The following codecs are supported by rtpengine:

  • G.711 (a-Law and µ-Law)
  • G.722
  • G.723.1
  • G.729
  • Speex
  • GSM
  • iLBC
  • Opus
  • AMR (narrowband and wideband)
  • EVS (if supplied -- see below)

Codec support is dependent on support provided by the ffmpeg codec libraries, which may vary from version to version. Use the --codecs command line option to have rtpengine print a list of codecs and their supported status. The list includes some codecs that are not listed above. Some of these are not actual VoIP codecs (such as MP3), while others lack support for encoding by ffmpeg at the time of writing (such as QCELP or ATRAC). If encoding support for these codecs becomes available in ffmpeg, rtpengine will be able to support them.

Audio format conversion including resampling and mono/stereo up/down-mixing happens automatically as required by the codecs involved. For example, one side could be using stereo Opus at 48 kHz sampling rate, and the other side could be using mono G.711 at 8 kHz, and rtpengine will perform the necessary conversions.

If repacketization (using the ptime option) is requested, the transcoding feature will also be engaged for the call, even if no additional codecs were requested.

G.729 support

As ffmpeg does not currently provide an encoder for G.729, transcoding support for it is available via the bcg729 library (mirror on GitHub). The build system looks for the bcg729 headers in a few locations and uses the library if found. If the library is located elsewhere, see daemon/Makefile to control where the build system is looking for it.

In a Debian build environment, debian/control lists a build-time dependency on bcg729. Newer Debian releases (currently bullseye, bookworm, sid) include bcg729 as a package so nothing needs to be done there. Older Debian releases do not currently include a bcg729 package, but one can be built locally using these instructions on GitHub. Sipwise provides a pre-packaged version of this as part of our C5 CE product which is available here.

Alternatively the build dependency can be removed from debian/control or by switching to a different Debian build profile. Set the environment variable export DEB_BUILD_PROFILES="pkg.ngcp-rtpengine.nobcg729" (or use the -P flag to the dpkg tools) and then build the rtpengine packages.

DTMF transcoding

Rtpengine supports transcoding between RFC 2833/4733 DTMF event packets (telephone-event payloads) and in-band DTMF audio tones. When enabled, rtpengine translates DTMF event packets to in-band DTMF audio by generating DTMF tones and injecting them into the audio stream, and translates in-band DTMF tones by running the audio stream through a DSP, and generating DTMF event packets when a DTMF tone is detected.

Support for DTMF transcoding can be enabled in one of two ways:

  • In the forward direction, DTMF transcoding is enabled by adding the codec telephone-event to the list of codecs offered for transcoding. Specifically, if the incoming SDP body doesn't yet list telephone-event as a supported codec, adding the option codec → transcode → telephone-event would enable DTMF transcoding. The receiving RTP client can then accept this codec and start sending DTMF event packets, which rtpengine would translate into in-band DTMF audio. If the receiving RTP client also offers telephone-event in their behalf, rtpengine would then detect in-band DTMF audio coming from the originating RTP client and translate it to DTMF event packets.

  • In the reverse direction, DTMF transcoding is enabled by adding the option always transcode to the flags if the incoming SDP body offers telephone-event as a supported codec. If the receiving RTP client then rejects the offered telephone-event codec, DTMF transcoding is then enabled and is performed in the same way as described above.

Enabling DTMF transcoding (in one of the two ways described above) implicitly enables the flag always transcode for the call and forces all of the audio to pass through the transcoding engine. Therefore, for performance reasons, this should only be done when really necessary.


Rtpengine can translate between fax endpoints that speak T.38 over UDPTL and fax endpoints that speak T.30 over regular audio channels. Any audio codec can theoretically be used for T.30 transmissions, but codecs that are too compressed will make the fax transmission fail. The most commonly used audio codecs for fax are the G.711 codecs (PCMU and PCMA), which are the default codecs rtpengine will use in this case if no other codecs are specified.

For further information, see the section on the T.38 dictionary key below.


As AMR supports dynamically adapting the encoder bitrate, as well as restricting the available bitrates, there are some slight peculiarities about its usage when transcoding.

When setting the bitrate, for example as AMR-WB/16000/1/23850 in either the codec-transcode or the codec-set options, that bitrate will be used as the highest permitted bitrate for the encoder. If no mode-set parameter is communicated in the SDP, then that is the bitrate that will be used.

If a mode-set is present, then the highest bitrate from that mode set which is lower or equal to the given bitrate will be used. If only higher bitrates are allowed by the mode set, then the next higher bitrate will be used.

To produce an SDP that includes the mode-set option (when adding AMR to the codec list via codec-transcode), the full format parameter string can be appended to the codec specification, e.g. codec-transcode-AMR-WB/16000/1/23850//mode-set=0,1,2,3,4,5;octet-align=1. In this example, the bitrate 23850 won't actually be used, as the highest permitted mode is 5 (18250 bps) and so that bitrate will be used.

If a literal = cannot be used due to parsing constraints (i.e. being wrongly interpreted as a key-value pair), it can be escaped by using two dashes instead, e.g. codec-transcode-AMR-WB/16000/1/23850//mode-set--0,1,2,3,4,5;octet-align--1

The default (highest) bitrates for AMR and AMR-WB are 6700 and 14250, respectively.

If a Codec Mode Request (CMR) is received from the AMR peer, then rtpengine will adhere to the request and switch encoder bitrate unconditionally, even if it's a higher bitrate than originally desired.

To enable sending CMRs to the AMR peer, the codec-specific option CMR-interval is provided. It takes a number of milliseconds as argument. Throughout each interval, rtpengine will track which AMR frame types were received from the peer, and then based on that will make a decision at the end of the interval. If a higher bitrate is allowed by the mode set that was not received from the AMR peer at all, then rtpengine will request switching to that bitrate per CMR. Only the next-highest bitrate mode that was not received will ever be requested, and a CMR will be sent only once per interval. Full example to specify a CMR interval of 500 milliseconds (with = escapes): codec-transcode-AMR-WB/16000/1/23850//mode-set--0,1,2/CMR-interval--500

Similar to the CMR-interval option, rtpengine can optionally attempt to periodically increase the outgoing bitrate without being requested to by the peer via a CMR. To enable this, set the option mode-change-interval to the desired interval in milliseconds. If the last CMR from the AMR peer was longer than this interval ago, rtpengine will increase the bitrate by one step if possible. Afterwards, the interval starts over.


Enhanced Voice Services (EVS) is a patent-encumbered codec for which (at the time of writing) no implementation exists which can be freely used and distributed. As such, support for EVS is only available if an implementation is supplied separately. Currently the only implementation supported is the ETSI/3GPP reference implementation (either floating-point or fixed-point). Any licensing issues that might result from such usage are the responsibility of the user of this software.

The EVS codec implementation can be provided as a shared object library (.so) which is loaded in during runtime (at startup). The supported implementations can be seen as subdirectories within the evs/ directory. Currently supported are version 17.0.0 of the ETSI/3GPP reference implementation, 126.442 for the fixed-point implementation and 126.443 for the floating-point implementation. (The floating-point implementation seems to be significantly faster, but is not bit-precise.)

To supply the codec implementation as a shared object during runtime, extract the reference implementation's .zip file and apply the provided patch (from here) that is appropriate for the chosen implementation. Run the build using make (suggested build flags are RELEASE=1 make) and it should produce a file Point rtpengine to this file using the evs-lib-path= option to enable support for EVS.

Call recording

Call recording can be accomplished in one of two ways:

  • The rtpengine daemon can write libpcap-formatted captures directly (--recording-method=pcap);

  • The rtpengine daemon can write audio frames into a sink in /proc/rtpengine (--recording-method=proc). These frames must then be consumed within a short period by another process; while this can be any process, the packaged rtpengine-recording daemon is a useful ready implementation of a call recording solution. The recording daemon uses ffmpeg libraries to implement a variety of on-the-fly format conversion and mixing options, as well as metadata logging. See rtpengine-recording -h for details.

Important note: The rtpengine daemon emits data into a "spool directory" (--recording-dir option), by default /var/spool/rtpengine. The recording daemon is then configured to consume this using the --spool-dir option, and to store the final emitted recordings (in whatever desired target format, etc.) in --output-dir. Ensure that the --spool-dir and the --output-dir are different directories, or you will run into problems (as discussed in #81).

The ng Control Protocol

In order to enable several advanced features in rtpengine, a new advanced control protocol has been devised which passes the complete SDP body from the SIP proxy to the rtpengine daemon, has the body rewritten in the daemon, and then passed back to the SIP proxy to embed into the SIP message.

This control protocol is supported over a number of different transports (plain UDP, plain TCP, HTTP, WebSocket) and loosely follows the same format as used by Kamailio's rtpproxy module. Each message passed between the SIP proxy and the media proxy contains of two parts: a unique message cookie and a dictionary document, separated by a single space. The message cookie is used to match requests to responses and to detect retransmissions. The message cookie in the response generated to a particular request therefore must be the same as in the request.

The dictionary document can be in one of two formats. It can be a JSON object or it can be a dictionary in bencode format. Bencoding supports a subset of the features of JSON (dictionaries/hashes, lists/arrays, arbitrary byte strings) but offers some benefits over JSON encoding, e.g. simpler and more efficient encoding, less encoding overhead, deterministic encoding and faster encoding and decoding. Disadvantages compared to JSON are that it's not a readily human readable format and that support in programming languages might be difficult to come by. Internally rtpengine uses bencoding natively, leading to additional overhead when JSON is in use as it has to be converted.

The dictionary of each request must contain at least one key called command. The corresponding value must be a string and determines the type of message. Currently the following commands are defined:

  • ping
  • offer
  • answer
  • delete
  • query
  • start recording
  • stop recording
  • block DTMF
  • unblock DTMF
  • block media
  • unblock media
  • silence media
  • unsilence media
  • start forwarding
  • stop forwarding
  • play media
  • stop media
  • play DTMF
  • statistics
  • publish
  • subscribe request
  • subscribe answer
  • unsubscribe

The response dictionary must contain at least one key called result. The value can be either ok or error. For the ping command, the additional value pong is allowed. If the result is error, then another key error-reason must be given, containing a string with a human-readable error message. No other keys should be present in the error case. If the result is ok, the optional key warning may be present, containing a human-readable warning message. This can be used for non-fatal errors.

For readability, all data objects below are represented in a JSON-like notation and without the message cookie. For example, a ping message and its corresponding pong reply would be written as:

{ "command": "ping" }
{ "result": "pong" }

While the actual messages as encoded on the wire, including the message cookie, might look like this in bencode format:

5323_1 d7:command4:pinge
5323_1 d6:result4:ponge

All keys and values are case-sensitive unless specified otherwise. The requirement stipulated by the bencode standard that dictionary keys must be present in lexicographical order is not currently honoured.

The ng protocol is used by Kamailio's rtpengine module, which is based on the older module called rtpproxy-ng, and utilises bencoding and the UDP transport by default, or alternatively WebSocket if so configured.

Of course the agent controlling rtpengine via the ng protocol does not have to be a SIP proxy. Any process that involves SDP can potentially talk to rtpengine via this protocol.

ping Message

The request dictionary contains no other keys and the reply dictionary also contains no other keys. The only valid value for result is pong.

offer Message

The request dictionary must contain at least the following keys:

  • sdp

    Contains the complete SDP body as string.

  • call-id

    The SIP call ID as string.

  • from-tag

    The SIP From tag as string.

Optionally included keys are:

  • from-tags

    Contains a list of strings used to selected multiple existing call participants (e.g. for the subscribe request message). An alternative way to list multiple tags is by putting them into the flags list, each prefixed with from-tags-.

  • via-branch

    The SIP Via branch as string. Used to additionally refine the matching logic between media streams and calls and call branches.

  • label or from-label

    A custom free-form string which rtpengine remembers for this participating endpoint and reports back in logs and statistics output. For some commands (e.g. block media) the given label is not used to set the label of the call participant, but rather to select an existing call participant.

  • set-label

    Some commands (e.g. block media) use the given label to select an existing call participant. For these commands, set-label instead of label can be used to set the label at the same time, either for the selected call participant (if selected via from-tag) or for the newly created participant (e.g. for subscribe request).

  • to-label

    Commands that allow selection of two call participants (e.g. block media) can use label instead of from-tag to select the first call participant. The to-label can then be used instead of to-tag to select the other call participant.

    For subscribe request the to-label is synonymous with set-label.

  • flags

    The value of the flags key is a list. The list contains zero or more of the following strings. Spaces in each string may be replaced by hyphens.

    • SIP source address

      Ignore any IP addresses given in the SDP body and use the source address of the received SIP message (given in received from) as default endpoint address. This was the default behaviour of older versions of rtpengine and can still be made the default behaviour through the --sip-source CLI switch. Can be overridden through the media address key.

    • trust address

      The opposite of SIP source address. This is the default behaviour unless the CLI switch --sip-source is active. Corresponds to the rtpproxy r flag. Can be overridden through the media address key.

    • symmetric

      Corresponds to the rtpproxy w flag. Not used by rtpengine as this is the default, unless asymmetric is specified.

    • asymmetric

      Corresponds to the rtpproxy a flag. Advertises an RTP endpoint which uses asymmetric RTP, which disables learning of endpoint addresses (see below).

    • unidirectional

      When this flag is present, kernelize also one-way rtp media.

    • strict source

      Normally, rtpengine attempts to learn the correct endpoint address for every stream during the first few seconds after signalling by observing the source address and port of incoming packets (unless asymmetric is specified). Afterwards, source address and port of incoming packets are normally ignored and packets are forwarded regardless of where they're coming from. With the strict source option set, rtpengine will continue to inspect the source address and port of incoming packets after the learning phase and compare them with the endpoint address that has been learned before. If there's a mismatch, the packet will be dropped and not forwarded.

    • media handover

      Similar to the strict source option, but instead of dropping packets when the source address or port don't match, the endpoint address will be re-learned and moved to the new address. This allows endpoint addresses to change on the fly without going through signalling again. Note that this opens a security hole and potentially allows RTP streams to be hijacked, either partly or in whole.

    • reset

      This causes rtpengine to un-learn certain aspects of the RTP endpoints involved, such as support for ICE or support for SRTP. For example, if ICE=force is given, then rtpengine will initially offer ICE to the remote endpoint. However, if a subsequent answer from that same endpoint indicates that it doesn't support ICE, then no more ICE offers will be made towards that endpoint, even if ICE=force is still specified. With the reset flag given, this aspect will be un-learned and rtpengine will again offer ICE to this endpoint. This flag is valid only in an offer message and is useful when the call has been transferred to a new endpoint without change of From or To tags.

    • port latching

      Forces rtpengine to retain its local ports during a signalling exchange even when the remote endpoint changes its port.

    • no port latching

      Port latching is enabled by default for endpoints which speak ICE. With this option preset, a remote port change will result in a local port change even for endpoints which speak ICE, which will imply an ICE restart.

    • record call

      Identical to setting record call to on (see below).

    • no rtcp attribute

      Omit the a=rtcp line from the outgoing SDP.

    • full rtcp attribute

      Include the full version of the a=rtcp line (complete with network address) instead of the short version with just the port number.

    • loop protect

      Inserts a custom attribute (a=rtpengine:...) into the outgoing SDP to prevent rtpengine processing and rewriting the same SDP multiple times. This is useful if your setup involves signalling loops and need to make sure that rtpengine doesn't start looping media packets back to itself. When this flag is present and rtpengine sees a matching attribute already present in the SDP, it will leave the SDP untouched and not process the message.

    • always transcode

      Legacy flag, synonymous to codec-accept=all.

    • single codec

      Using this flag in an answer message will leave only the first listed codec in place and will remove all others from the list. Useful for RTP clients which get confused if more than one codec is listed in an answer.

    • reuse codecs or no codec renegotiation

      Instructs rtpengine to prevent endpoints from switching codecs during call run-time if possible. Codecs that were listed as preferred in the past will be kept as preferred even if the re-offer lists other codecs as preferred, or in a different order. Recommended to be combined with single codec.

    • allow transcoding

      This flag is only useful in commands that provide an explicit answer SDP to rtpengine (e.g. subscribe answer). For these commands, if the answer SDP does not accept all codecs that were offered, the default behaviour is to reject the answer. With this flag given, the answer will be accepted even if some codecs were rejected, and codecs will be transcoded as required.

    • all

      Synonymous to all=all (see below).

    • pad crypto

      Legacy alias to SDES=pad.

    • generate mid

      Add a=mid attributes to the outgoing SDP if they were not already present.

    • strip extmap

      Remove a=rtpmap attributes from the outgoing SDP.

    • original sendrecv

      With this flag present, rtpengine will leave the media direction attributes (sendrecv, recvonly, sendonly, and inactive) from the received SDP body unchanged. Normally rtpengine would consume these attributes and insert its own version of them based on other media parameters (e.g. a media section with a zero IP address would come out as sendonly or inactive).

    • inject DTMF

      Signals to rtpengine that the audio streams involved in this offer or answer (the flag should be present in both of them) are to be made available for DTMF injection via the play DTMF control message. See play DTMF below for additional information.

    • detect DTMF

      When present in a message that sets up codec handlers, enables the DSP to detect in-band DTMF audio tones even when it wouldn't otherwise be necessary.

    • generate RTCP

      Identical to setting generate RTCP = on.

    • RTCP mirror

      Useful only for subscribe request message. Instructs rtpengine to not only create a one-way subscription for both RTP and RTCP from the source to the sink, but also create a reverse subscription for RTCP only from the sink back to the source. This makes it possible for the media source to receive feedback from all media receivers (sinks).

    • debug or debugging

      Enabled full debug logging for this call, regardless of global log level settings.

    • pierce NAT

      Sends empty UDP packets to the remote RTP peer as soon as an endpoint address is available from a received SDP, for as long as no incoming packets have been received. Useful to create an initial NAT mapping. Not needed when ICE is in use.

    • NAT-wait

      Prevents forwarding media packets to the respective endpoint until at least one media packet has been received from that endpoint. This is to allow a NAT binding to open in the ingress direction before sending packets out, which could result in an automated firewall block.

    • trickle ICE

      Useful for offer messages when ICE is advertised to also advertise support for trickle ICE.

    • reject ICE

      Useful for offer messages that advertise support for ICE. Instructs rtpengine to reject the offered ICE. This is similar to using ICE=remove in the respective answer.

  • generate RTCP

    Contains a string, either on or off. If enabled for a call, received RTCP packets will not simply be passed through as usual, but instead will be consumed, and instead rtpengine will generate its own RTCP packets to send to the RTP peers. This flag will be effective for both sides of a call.

  • replace

    Similar to the flags list. Controls which parts of the SDP body should be rewritten. Contains zero or more of:

    • origin

      Replace the address found in the origin (o=) line of the SDP body. Corresponds to rtpproxy o flag.

    • session connection or session-connection

      Replace the address found in the session-level connection (c=) line of the SDP body. Corresponds to rtpproxy c flag.

    • SDP version or SDP-version

      Take control of the version field in the SDP and make sure it's increased every time the SDP changes, and left unchanged if the SDP is the same.

    • username

      Take control of the origin username field in the SDP. With this option in use, rtpengine will make sure the username field in the o= line always remains the same in all SDPs going to a particular RTP endpoint.

    • session name or session-name

      Same as username but for the entire contents of the s= line.

    • zero address

      Using a zero endpoint address is an obsolete way to signal a muted or sendonly stream. Streams with zero addresses are normally flagged as sendonly and the zero address in the SDP is passed through. With this option set, the zero address is replaced with a real address.

  • direction

    Contains a list of two strings and corresponds to the rtpproxy e and i flags. Each element must correspond to one of the named logical interfaces configured on the command line (through --interface). For example, if there is one logical interface named pub and another one named priv, then if side A (originator of the message) is considered to be on the private network and side B (destination of the message) on the public network, then that would be rendered within the dictionary as:

      { ..., "direction": [ "priv", "pub" ], ... }

    This only needs to be done for an initial offer; for the answer and any subsequent offers (between the same endpoints) rtpengine will remember the selected network interface.

    As a special case to support legacy usage of this option, if the given interface names are internal or external and if no such interfaces have been configured, then they're understood as selectors between IPv4 and IPv6 addresses. However, this mechanism for selecting the address family is now obsolete and the address family dictionary key should be used instead.

    For legacy support, the special direction keyword round-robin-calls can be used to invoke the round-robin interface selection algorithm described in the section Interfaces configuration. If this special keyword is used, the round-robin selection will run over all configured interfaces, whether or not they are configured using the BASE:SUFFIX interface name notation. This special keyword is provided only for legacy support and should be considered obsolete. It will be removed in future versions.

  • interface

    Contains a single string naming one of the configured interfaces, just like direction does. The interface option is used instead of direction where only one interface is required (e.g. outside of an offer/answer scenario), for example in the publish or subscribe request commands.

  • received from

    Contains a list of exactly two elements. The first element denotes the address family and the second element is the SIP message's source address itself. The address family can be one of IP4 or IP6. Used if SDP addresses are neither trusted (through SIP source address or --sip-source) nor the media address key is present.

  • drop-traffic

    Contains a string, valid values are start or stop.

    start signals to rtpengine that all RTP involved in this call is dropped. Can be present either in offer or answer, the behavior is for the entire call.

    stop signals to rtpengine that all RTP involved in this call is NOT dropped anymore. Can be present either in offer or answer, the behavior is for the entire call.

    stop has priority over start, if both are present.

  • ICE

    Contains a string which must be one of the following values:

    With remove, any ICE attributes are stripped from the SDP body. Also see the flag reject ICE to effect an early removal of ICE support during an offer.

    With force, ICE attributes are first stripped, then new attributes are generated and inserted, which leaves the media proxy as the only ICE candidate.

    With default, the behaviour will be the same as with force if the incoming SDP already had ICE attributes listed. If the incoming SDP did not contain ICE attributes, then no ICE attributes are added.

    With force-relay, existing ICE candidates are left in place except relay type candidates, and rtpengine inserts itself as a relay candidate. It will also leave SDP c= and m= lines unchanged.

    With optional, if no ICE attributes are present, a new set is generated and the media proxy lists itself as ICE candidate; otherwise, the media proxy inserts itself as a low-priority candidate. This used to be the default behaviour in previous versions of rtpengine.

    The default behaviour (no ICE key present at all) is the same as default.

    This flag operates independently of the replace flags.

    Note that if config parameter save-interface-ports = true, ICE will be broken, because rtpengine will bind ports only on the first local interface of desired family of logical interface.

  • ICE-lite

    Contains a string which must be one of the following values:

    • forward to enable "ICE lite" mode towards the peer that this offer is sent to.

    • backward to enable "ICE lite" mode towards the peer that has sent this offer.

    • both to enable "ICE lite" towards both peers.

    • off to disable "ICE lite" towards both peers and revert to full ICE support.

    The default (keyword not present at all) is to use full ICE support, or to leave the previously set "ICE lite" mode unchanged. This keyword is valid in offer messages only.

  • transport protocol

    The transport protocol specified in the SDP body is to be rewritten to the string value given here. The media proxy will expect to receive this protocol on the allocated ports, and will talk this protocol when sending packets out. Translation between different transport protocols will happen as necessary.

    Valid values are: RTP/AVP, RTP/AVPF, RTP/SAVP, RTP/SAVPF.

    Additionally the string accept can be given in answer messages to allow a special case: By default (when no transport-protocol override is given) in answer messages, rtpengine will use the transport protocol that was originally offered. However, an answering client may answer with a different protocol than what was offered (e.g. offer was for RTP/AVP and answer comes with RTP/AVPF). The default behaviour for rtpengine is to ignore this protocol change and still proceed with the protocol that was originally offered. Using the accept option here tells rtpengine to go along with this protocol change and pass it to the original offerer.

  • media address

    This can be used to override both the addresses present in the SDP body and the received from address. Contains either an IPv4 or an IPv6 address, expressed as a simple string. The format must be dotted-quad notation for IPv4 or RFC 5952 notation for IPv6. It's up to the RTP proxy to determine the address family type.

  • address family

    A string value of either IP4 or IP6 to select the primary address family in the substituted SDP body. The default is to auto-detect the address family if possible (if the receiving end is known already) or otherwise to leave it unchanged.

  • rtcp-mux

    A list of strings controlling the behaviour regarding rtcp-mux (multiplexing RTP and RTCP on a single port, RFC 5761). The default behaviour is to go along with the client's preference. The list can contain zero of more of the following strings. Note that some of them are mutually exclusive.

    • offer

      Instructs rtpengine to always offer rtcp-mux, even if the client itself doesn't offer it.

    • require

      Similar to offer but pretends that the receiving client has already accepted rtcp-mux. The effect is that no separate RTCP ports will be advertised, even in an initial offer (which is against RFC 5761). This option is provided to talk to WebRTC clients.

    • demux

      If the client is offering rtcp-mux, don't offer it to the other side, but accept it back to the offering client.

    • accept

      Instructs rtpengine to accept rtcp-mux and also offer it to the other side if it has been offered.

    • reject

      Reject rtcp-mux if it has been offered. Can be used together with offer to achieve the opposite effect of demux.

  • TOS

    Contains an integer. If present, changes the TOS value for the entire call, i.e. the TOS value used in outgoing RTP packets of all RTP streams in all directions. If a negative value is used, the previously used TOS value is left unchanged. If this key is not present or its value is too large (256 or more), then the TOS value is reverted to the default (as per --tos command line).

  • DTLS

    Contains a string and influences the behaviour of DTLS-SRTP. Possible values are:

    • off or no or disable

      Prevents rtpengine from offering or acceping DTLS-SRTP when otherwise it would. The default is to offer DTLS-SRTP when encryption is desired and to favour it over SDES when accepting an offer.

    • passive

      Instructs rtpengine to prefer the passive (i.e. server) role for the DTLS handshake. The default is to take the active (client) role if possible. This is useful in cases where the SRTP endpoint isn't able to receive or process the DTLS handshake packets, for example when it's behind NAT or needs to finish ICE processing first.

    • active

      Reverts the passive setting. Only useful if the dtls-passive config option is set.

  • DTLS-reverse

    Contains a string and influences the behaviour of DTLS-SRTP. Unlike the regular DTLS flag, this one is used to control behaviour towards DTLS that was offered to rtpengine. In particular, if passive mode is used, it prevents rtpengine from prematurely sending active DTLS connection attempts. Possible values are:

    • passive

      Instructs rtpengine to prefer the passive (i.e. server) role for the DTLS handshake. The default is to take the active (client) role if possible. This is useful in cases where the SRTP endpoint isn't able to receive or process the DTLS handshake packets, for example when it's behind NAT or needs to finish ICE processing first.

    • active

      Reverts the passive setting. Only useful if the dtls-passive config option is set.

  • DTLS-fingerprint

    Contains a string and is used to select the hashing function to generate the DTLS fingerprint from the certificate. The default is SHA-256, or the same hashing function as was used by the peer. Available are SHA-1, SHA-224, SHA-256, SHA-384, and SHA-512.

  • SDES

    A list of strings controlling the behaviour regarding SDES. The default is to offer SDES without any session parameters when encryption is desired, and to accept it when DTLS-SRTP is unavailable. If two SDES endpoints are connected to each other, then the default is to offer SDES with the same options as were received from the other endpoint. Additionally, all other supported SDES crypto suites are added to the outgoing offer by default.

    These options can also be put into the flags list using a prefix of SDES-. All options controlling SDES session parameters can be used either in all lower case or in all upper case.

    • off or no or disable

      Prevents rtpengine from offering SDES, leaving DTLS-SRTP as the other option.

    • unencrypted_srtp, unencrypted_srtcp and unauthenticated_srtp

      Enables the respective SDES session parameter (see section 6.3 or RFC 4568). The default is to copy these options from the offering client, or not to have them enabled if SDES wasn't offered.

    • encrypted_srtp, encrypted_srtcp and authenticated_srtp

      Negates the respective option. This is useful if one of the session parameters was offered by an SDES endpoint, but it should not be offered on the far side if this endpoint also speaks SDES.

    • no-SUITE

      Exclude individual crypto suites from being included in the offer. For example, no-NULL_HMAC_SHA1_32 would exclude the crypto suite NULL_HMAC_SHA1_32 from the offer. This has two effects: if a given crypto suite was present in a received offer, it will be removed and will be missing in the outgoing offer; and if a given crypto suite was not present in the received offer, it will not be added to it.

    • pad

      RFC 4568 (section 6.1) is somewhat ambiguous regarding the base64 encoding format of a=crypto parameters added to an SDP body. The default interpretation is that trailing = characters used for padding should be omitted. With this flag set, these padding characters will be left in place.

    • lifetime

      Add the key lifetime parameter 2^31 to each crypto key.

    • static

      Instructs rtpengine to skip the full SDES negotiation routine during a re-invite (e.g. pick the first support crypto suite, look for possible SRTP passthrough) and instead leave the previously negotiated crypto suite in place. Only useful in subsequent answer messages and ignored in offer messages.


    Similar to SDES but controls OSRTP behaviour. Default behaviour is to pass through OSRTP negotiations. Supported options:

    • offer

      When processing a non-OSRTP offer, convert it to an OSRTP offer. Will result in RTP/SRTP transcoding if the OSRTP offer is accepted.

    • accept

      When processing a non-OSRTP answer in response to an OSRTP offer, accept the OSRTP offer anyway. Results in RTP/SRTP transcoding.

  • endpoint-learning

    Contains one of the strings off, immediate, delayed or heuristic. This tells rtpengine which endpoint learning algorithm to use and overrides the endpoint-learning configuration option. This option can also be put into the flags list using a prefix of endpoint-learning-.

  • record call

    Contains one of the strings yes, no, on or off. This tells the rtpengine whether or not to record the call to PCAP files. If the call is recorded, it will generate PCAP files for each stream and a metadata file for each call. Note that rtpengine will not force itself into the media path, and other flags like ICE=force may be necessary to ensure the call is recorded.

    See the --recording-dir option above.

    Enabling call recording via this option has the same effect as doing it separately via the start recording message, except that this option guarantees that the entirety of the call gets recorded, including all details such as SDP bodies passing through rtpengine.

  • metadata

    This is a generic metadata string. The metadata will be written to the bottom of metadata files within /path/to/recording_dir/metadata/ or to recording_metakeys table. In the latter case, metadata string must contain a list of key:val pairs separated by | character. metadata can be used to record additional information about recorded calls. metadata values passed in through subsequent messages will overwrite previous metadata values.

    See the --recording-dir option above.

  • codec

    Contains a dictionary controlling various aspects of codecs (or RTP payload types).

    These options can also be put into the flags list using a prefix of codec-. For example, to set the codec options for two variants of Opus when they're implicitly accepted, (see the example under set), one would put the following into the flags list: codec-set-opus/48000/1/16000 codec-set-opus/48000/2/32000

    The following keys are understood:

    • strip

      Contains a list of strings. Each string is the name of a codec or RTP payload type that should be removed from the SDP. Codec names are case sensitive, and can be either from the list of codecs explicitly defined by the SDP through an a=rtpmap attribute, or can be from the list of RFC-defined codecs. Examples are PCMU, opus, or telephone-event. Codecs stripped using this option are treated as if they had never been in the SDP.

      It is possible to specify codec format parameters alongside with the codec name in the same format as they're written in SDP for codecs that support them, for example opus/48000 to specify Opus with 48 kHz sampling rate and one channel (mono), or opus/48000/2 for stereo Opus. If any format parameters are specified, the codec will only be stripped if all of the format parameters match, and other instances of the same codec with different format parameters will be left untouched.

      As a special keyword, all can be used to remove all codecs, except the ones that should explicitly offered (see below). Note that it is an error to strip all codecs and leave none that could be offered. In this case, the original list of codecs will be left unchanged.

      The keyword full can also be used, which behaves the same as all with the exception listed under transcode below.

    • except

      Contains a list of strings. Each string is the name of a codec that should be included in the list of codecs offered. This is primarily useful to block all codecs (strip -> all or mask -> all) except the ones given in the except whitelist. Codecs that were not present in the original list of codecs offered by the client will be ignored.

      This list also supports codec format parameters as per above.

    • offer

      This is identical to except but additionally allows the codec order to be changed. So the first codec listed in offer will be the primary (preferred) codec in the output SDP, even if it wasn't originally so.

    • transcode

      Similar to offer but allows codecs to be added to the list of offered codecs even if they were not present in the original list of codecs. In this case, the transcoding engine will be engaged. Only codecs that are supported for both decoding and encoding can be added in this manner. This also has the side effect of automatically stripping all unsupported codecs from the list of offered codecs, as rtpengine must expect to receive or even send in any codec that is present in the list.

      Note that using this option does not necessarily always engage the transcoding engine. If all codecs given in the transcode list were present in the original list of offered codecs, then no transcoding will be done. Also note that if transcoding takes place, in-kernel forwarding is disabled for this media stream and all processing happens in userspace.

      If no codec format parameters are specified in this list (e.g. just opus instead of opus/48000/2), default values will be chosen for them.

      For codecs that support different bitrates, it can be specified by appending another slash followed by the bitrate in bits per second, e.g. opus/48000/2/32000. In this case, all format parameters (clock rate, channels) must also be specified.

      Additional options that can be appended to the codec string with additional slashes are ptime, the fmtp string, and additional codec-specific options. For example iLBC/8000/1///mode=30 to use mode=30 as fmtp string.

      For Opus, the string of codec-specific options is passed directly to ffmpeg, so all ffmpeg codec options can be set. Use space, colon, semicolon, or comma to separate individual options. For example to set the encoding complexity (also known as compression level by ffmpeg): opus/48000/2////compression_level=2

      If a literal = cannot be used due to parsing constraints (i.e. being wrongly interpreted as a key-value pair), it can be escaped by using two dashes instead, e.g. iLBC/8000/1///mode--30.

      As a special case, if the strip=all or mask=all option has been used and the transcode option is used on a codec that was originally present in the offer, then rtpengine will treat this codec the same as if it had been used with the offer option, i.e. it will simply restore it from the list of stripped codecs and won't actually engage transcoding for this codec. On the other hand, if a codec has been stripped explicitly by name using the strip or mask option and then used again with the transcode option, then the codec will not simply be restored from the list of stripped codecs, but instead a new transcoded instance of the codec will be inserted into the offer. (This special exception does not apply to mask=full or strip=full.)

      This option is only processed in offer messages and ignored otherwise.

    • mask

      Similar to strip except that codecs listed here will still be accepted and used for transcoding on the offering side. Useful only in combination with transcode. For example, if an offer advertises Opus and the options mask=opus, transcode=G723 are given, then the rewritten outgoing offer will contain only G.723 as offered codec, and transcoding will happen between Opus and G.723. In contrast, if only transcode=G723 were given, then the rewritten outgoing offer would contain both Opus and G.723. On the other hand, if strip=opus, transcode=G723 were given, then Opus would be unavailable for transcoding.

      As with the strip option, the special keywords all and full can be used to mask all codecs that have been offered.

      This option is only processed in offer messages and ignored otherwise.

    • consume

      Identical to mask but enables the transcoding engine even if no other transcoding related options are given.

    • accept

      Similar to mask and consume but doesn't remove the codec from the list of offered codecs. This means that a codec listed under accept will still be offered to the remote peer, but if the remote peer rejects it, it will still be accepted towards the original offerer and then used for transcoding. It is a more selective version of what the always transcode flag does.

      The special string any can be used for the publish message. See below for more details.

    • set

      Contains a list of strings. This list makes it possible to set codec options (bitrate in particular) for codecs that are implicitly accepted for transcoding. For example, if AMR was offered, transcode=PCMU was given, and the remote ended up accepting PCMU, then this option can be used to set the bitrate used for the AMR transcoding process.

      Each string must be a full codec specification as per above, including clock rate and number of channels. Using the example above, set=AMR/8000/1/7400 can be used to transcode to AMR with 7.4 kbit/s.

      Codec options (bitrate) are only applied to codecs that match the given parameters (clock rate, channels), and multiple options can be given for the same coded with different parameters. For example, to specify different bitrates for Opus for both mono and stereo output, one could use set=[opus/48000/1/16000,opus/48000/2/32000].

      This option is only processed in offer messages and ignored otherwise.

  • ptime

    Contains an integer. If set, changes the a=ptime attribute's value in the outgoing SDP to the provided value. It also engages the transcoding engine for supported codecs to provide repacketization functionality, even if no additional codec has actually been requested for transcoding. Note that not all codecs support all packetization intervals.

    The selected ptime (which represents the duration of a single media packet in milliseconds) will be used towards the endpoint receiving this offer, even if the matching answer prefers a different ptime.

    This option is ignored in answer messages. See below for the reverse.

  • ptime-reverse

    This is the reciprocal to ptime. It sets the ptime to be used towards the endpoint who has sent the offer. It will be inserted in the answer SDP. This option is also ignored in answer messages.

  • T.38

    Contains a list of strings. Each string is a flag that controls the behaviour regarding T.38 transcoding. These flags are ignored if the message is not an offer. Recognised flags are:

    • decode

      If the received SDP contains a media section with an image type, UDPTL transport, and t38 format string, this flag instructs rtpengine to convert this media section into an audio type using RTP as transport protocol. Other transport protocols (such as SRTP) can be selected using transport protocol as described above.

      The default audio codecs to be offered are PCMU and PCMA. Other audio codecs can be specified using the transcode= flag described above, in which case the default codecs will not be offered automatically.

    • force

      If the received SDP contains an audio media section using RTP transport, this flag instructs rtpengine to convert it to an image type media section using the UDPTL protocol. The first supported audio codec that was offered will be used to transport T.30. Default options for T.38 are used for the generated SDP.

    • stop

      Stops a currently active T.38 gateway that was previously engaged using the decode or force flags. This is useful to handle a rejected T.38 offer and revert the session back to media passthrough.

    • no-ECM

      Disable support for ECM. Support is enabled by default.

    • no-V.17

      Disable support for V.17. Support is enabled by default.

    • no-V.27ter

      Disable support for V.27ter. Support is enabled by default.

    • no-V.29

      Disable support for V.29. Support is enabled by default.

    • no-V.34

      Disable support for V.34. Support is enabled by default.

    • no-IAF

      Disable support for IAF. Support is enabled by default.

    • FEC

      Use UDPTL FEC instead of redundancy. Only useful with T.38=force as it's a negotiated parameter.

  • supports

    Contains a list of strings. Each string indicates support for an additional feature that the controlling SIP proxy supports. Currently defined values are:

    • load limit

      Indicates support for an extension to the ng protocol to facilitate certain load balancing mechanisms. If rtpengine is configured with certain session or load limit options enabled (such as the max-sessions option), then normally rtpengine would reply with an error to an offer if one of the limits is exceeded. If support for the load limit extension is indicated, then instead of replying with an error, rtpengine responds with the string load limit in the result key of the response dictionary. The response dictionary may also contain the optional key message with an explanatory string. No other key is required in the response dictionary.

  • xmlrpc-callback

    Contains a string that encodes an IP address (either IPv4 or IPv6) in printable format. If specified, then this address will be used as destination address for the XMLRPC timeout callback (see b2b-url option).

  • media echo or media-echo

    Contains a string to enable a special media echo mode. Recognised values are:

    • blackhole or sinkhole

      Media arriving from either side of the call is simply discarded and not forwarded.

    • forward

      Enables media echo towards the receiver of this message (e.g. the called party if the message is an offer from the caller). Media arriving from that side is echoed back to its sender (with a new SSRC if it's RTP). Media arriving from the opposite side is discarded.

    • backwards

      Enables media echo towards the sender of this message (i.e. the opposite of forward). Media arriving from the other side is discarded.

    • both

      Enables media echo towards both the sender and the receiver of this message.

  • DTMF-security

    Used in the block DTMF message to select the DTMF blocking mode. The default mode is drop which simply drops DTMF event packets. The other supported modes are: silence which replaces DTMF events with silence audio; tone which replaces DTMF events with a single sine wave tone; random which replaces DTMF events with random other DTMF events (both in-band DTMF audio tones and RFC event packets); zero which is similar to random except that a zero event is always used; DTMF which is similar to zero except that a different DTMF digit can be specified; off to disable DTMF blocking.

  • DTMF-security-trigger

    Blocking mode to enable when the DTMF trigger (see below) is detected.

  • DTMF-security-trigger-end

    Blocking mode to enable when the DTMF end trigger (see below) is detected.

  • trigger

    A string of DTMF digits that enable a DTMF blocking mode when detected.

  • end trigger or trigger-end

    A string of DTMF digits that disable DTMF blocking or enable a different DTMF blocking mode when detected, but only after the initial enabling trigger has been detected.

  • trigger-end-time

    Time in milliseconds that a DTMF blocking mode enabled by the trigger should remain active the most. After the time has expired, the blocking mode is switched to the trigger-end mode.

  • trigger-end-digits

    Number of DTMF digits that a DTMF blocking mode enabled by the trigger should remain active the most. After this number of DTMF digits has been detected, the blocking mode is switched to the trigger-end mode.

  • frequency

    Sets the tone frequency for DTMF-security=tone in Hertz. The default is 400 Hz.

  • volume

    Sets the tone volume for DTMF-security modes tone, zero, DTMF, and random` in negative dB. The default is -10 dB. The highest possible volume is 0 dB and the lowest possible volume is -63 dB.

  • digit or code

    Sets the replacement digit for DTMF-security=DTMF.

  • delay-buffer

    Takes an integer as value. When set to non-zero, enables the delay buffer when setting up codec handlers. The delay buffer delays all media by the given number of milliseconds before passing it on. Once the delay buffer is configured, it must explicitly be disabled again by setting this value to zero. The delay buffer setting is honoured in all messages that set up codec handlers, such as block DTMF.

  • DTMF-delay

    Time in milliseconds to delay DTMF events (both RFC event packets and DTMF tones) for. With this option enabled (set to non-zero), DTMF events are initially replaced by silence and then subsequently reproduced after the given delay. DTMF blocking modes are honoured at the time when the DTMF events are reproduced.

  • all

    Can be set to the string none to disable any extra behaviour (which is the default if this key is omitted altogether) or to one of all, offer-answer, except-offer-answer or flows. Applicable to certain messages only. The behaviour is explained below separately for each affected message.

An example of a complete offer request dictionary could be (SDP body abbreviated):

{ "command": "offer", "call-id": "cfBXzDSZqhYNcXM", "from-tag": "mS9rSAn0Cr",
"sdp": "v=0\r\no=...", "via-branch": "5KiTRPZHH1nL6",
"flags": [ "trust address" ], "replace": [ "origin", "session connection" ],
"address family": "IP6", "received-from": [ "IP4", "" ],
"ICE": "force", "transport protocol": "RTP/SAVPF", "media address": "2001:d8::6f24:65b",
"DTLS": "passive" }

The response message only contains the key sdp in addition to result, which contains the re-written SDP body that the SIP proxy should insert into the SIP message.

Example response:

{ "result": "ok", "sdp": "v=0\r\no=..." }

answer Message

The answer message is identical to the offer message, with the additional requirement that the dictionary must contain the key to-tag containing the SIP To tag. It doesn't make sense to include the direction key in the answer message.

The reply message is identical as in the offer reply.

delete Message

The delete message must contain at least the keys call-id and from-tag and may optionally include to-tag and via-branch, as defined above. It may also optionally include a key flags containing a list of zero or more strings. The following flags are defined:

  • fatal

    Specifies that any non-syntactical error encountered when deleting the stream (such as unknown call-ID) shall result in an error reply (i.e. "result": "error"). The default is to reply with a warning only (i.e. "result": "ok", "warning": ...).

Other optional keys are:

  • delete delay

    Contains an integer and overrides the global command-line option delete-delay. Call/branch will be deleted immediately if a zero is given. Value must be positive (in seconds) otherwise.

The reply message may contain additional keys with statistics about the deleted call. Those additional keys are the same as used in the query reply.

list Message

The list command retrieves the list of currently active call-ids. This list is limited to 32 elements by default.

  • limit

    Optional integer value that specifies the maximum number of results (default: 32). Must be > 0. Be careful when setting big values, as the response may not fit in a UDP packet, and therefore be invalid.

query Message

The minimum requirement is the presence of the call-id key. Keys from-tag and/or to-tag may optionally be specified.

The response dictionary contains the following keys:

  • created

    Contains an integer corresponding to the creation time of this call within the media proxy, expressed as seconds since the UNIX epoch.

  • last signal

    The last time a signalling event (offer, answer, etc) occurred. Also expressed as an integer UNIX timestamp.

  • tags

    Contains a dictionary. The keys of the dictionary are all the SIP tags (From-tag, To-Tag) known by rtpengine related to this call. One of the keys may be an empty string, which corresponds to one side of a dialogue which hasn't signalled its SIP tag yet. Each value of the dictionary is another dictionary with the following keys:

    • created

      UNIX timestamp of when this SIP tag was first seen by rtpengine.

    • tag

      Identical to the corresponding key of the tags dictionary. Provided to allow for easy traversing of the dictionary values without paying attention to the keys.

    • label

      The label assigned to this endpoint in the offer or answer message.

    • in dialogue with

      Contains the SIP tag of the other side of this dialogue. May be missing in case of a half-established dialogue, in which case the other side is represented by the null-string entry of the tags dictionary.

    • medias

      Contains a list of dictionaries, one for each SDP media stream known to rtpengine. The dictionaries contain the following keys:

      • index

        Integer, sequentially numbered index of the media, starting with one.

      • type

        Media type as string, usually audio or video.

      • protocol

        If the protocol is recognized by rtpengine, this string contains it. Usually RTP/AVP or RTP/SAVPF.

      • flags

        A list of strings containing various status flags. Contains zero of more of: initialized, rtcp-mux, DTLS-SRTP, SDES, passthrough, ICE.

      • streams

        Contains a list of dictionary representing the packet streams associated with this SDP media. Usually contains two entries, one for RTP and one for RTCP. The keys found in these dictionaries are listed below:

      • local port

        Integer representing the local UDP port. May be missing in case of an inactive stream.

      • endpoint

        Contains a dictionary with the keys family, address and port. Represents the endpoint address used for packet forwarding. The family may be one of IPv4 or IPv6.

      • advertised endpoint

        As above, but representing the endpoint address advertised in the SDP body.

      • crypto suite

        Contains a string such as AES_CM_128_HMAC_SHA1_80 representing the encryption in effect. Missing if no encryption is active.

      • last packet

        UNIX timestamp of when the last UDP packet was received on this port.

      • flags

        A list of strings with various internal flags. Contains zero or more of: RTP, RTCP, fallback RTCP, filled, confirmed, kernelized, no kernel support.

      • stats

        Contains a dictionary with the keys bytes, packets and errors. Statistics counters for this packet stream.

  • totals

    Contains a dictionary with two keys, RTP and RTCP, each one containing another dictionary identical to the stats dictionary described above.

A complete response message might look like this (formatted for readability):

        "totals": {
          "RTCP": {
                "bytes": 2244,
                "errors": 0,
                "packets": 22
          "RTP": {
               "bytes": 100287,
               "errors": 0,
               "packets": 705
        "last_signal": 1402064116,
        "tags": {
              "cs6kn1rloc": {
              "created": 1402064111,
              "medias": [
                  "flags": [
                  "streams": [
                       "endpoint": {
                           "port": 57370,
                           "address": "10.xx.xx.xx",
                           "family": "IPv4"
                       "flags": [
                       "local port": 30018,
                       "last packet": 1402064124,
                       "stats": {
                              "packets": 343,
                              "errors": 0,
                              "bytes": 56950
                       "advertised endpoint": {
                                "family": "IPv4",
                                "port": 57370,
                                "address": "10.xx.xx.xx"
                       "stats": {
                              "bytes": 164,
                              "errors": 0,
                              "packets": 2
                       "advertised endpoint": {
                                "family": "IPv4",
                                "port": 57371,
                                "address": "10.xx.xx.xx"
                       "endpoint": {
                           "address": "10.xx.xx.xx",
                           "port": 57371,
                           "family": "IPv4"
                       "last packet": 1402064123,
                       "local port": 30019,
                       "flags": [
                              "no kernel support"
                  "protocol": "RTP/AVP",
                  "index": 1,
                  "type": "audio"
              "in dialogue with": "0f0d2e18",
              "tag": "cs6kn1rloc"
              "0f0d2e18": {
                  "in dialogue with": "cs6kn1rloc",
                  "tag": "0f0d2e18",
                  "medias": [
                      "protocol": "RTP/SAVPF",
                      "index": 1,
                      "type": "audio",
                      "streams": [
                           "endpoint": {
                               "family": "IPv4",
                               "address": "10.xx.xx.xx",
                               "port": 58493
                           "crypto suite": "AES_CM_128_HMAC_SHA1_80",
                           "local port": 30016,
                           "last packet": 1402064124,
                           "flags": [
                           "stats": {
                            "bytes": 43337,
                            "errors": 0,
                            "packets": 362
                           "advertised endpoint": {
                              "address": "10.xx.xx.xx",
                              "port": 58493,
                              "family": "IPv4"
                           "local port": 30017,
                           "last packet": 1402064124,
                           "flags": [
                            "no kernel support"
                           "endpoint": {
                               "family": "IPv4",
                               "port": 60193,
                               "address": "10.xx.xx.xx"
                           "crypto suite": "AES_CM_128_HMAC_SHA1_80",
                           "advertised endpoint": {
                              "family": "IPv4",
                              "port": 60193,
                              "address": "10.xx.xx.xx"
                           "stats": {
                            "packets": 20,
                            "bytes": 2080,
                            "errors": 0
                      "flags": [
                  "created": 1402064111
        "created": 1402064111,
        "result": "ok"

start recording Message

The start recording message must contain at least the key call-id and may optionally include from-tag, to-tag and via-branch, as defined above. The reply dictionary contains no additional keys.

Enables call recording for the call, either for the entire call or for only the specified call leg. Currently rtpengine always enables recording for the entire call and does not support recording only individual call legs, therefore all keys other than call-id are currently ignored.

If the chosen recording method doesn't support in-kernel packet forwarding, enabling call recording via this messages will force packet forwarding to happen in userspace only.

If the optional 'output-destination' key is set, then its value will be used as an output file. Note that a filename extension will not be added.

stop recording Message

The stop recording message must contain the key call-id as defined above. The reply dictionary contains no additional keys.

Disables call recording for the call. This can be sent during a call to immediately stop recording it.

block DTMF and unblock DTMF Messages

These message types must include the key call-id in the message. They enable and disable blocking of DTMF events (RFC 4733 type packets), respectively.

Packets can be blocked for an entire call if only the call-id key is present in the message, or can be blocked directionally for individual participants. Participants can be selected by their SIP tag if the from-tag key is included in the message, they can be selected by their SDP media address if the address key is included in the message, or they can be selected by the user-provided label if the label key is included in the message. For an address, it can be an IPv4 or IPv6 address, and any participant that is found to have a matching address advertised as their SDP media address will have their originating RTP packets blocked (or unblocked).

Unblocking packets for the entire call (i.e. only call-id is given) does not automatically unblock packets for participants which had their packets blocked directionally, unless the string all (equivalent to setting all=all) is included in the flags section of the message.

When DTMF blocking is enabled, DTMF event packets will not be forwarded to the receiving peer. If DTMF logging is enabled, DTMF events will still be logged to syslog while blocking is enabled. Blocking of DTMF events can be enabled and disabled at any time during call runtime.

block media and unblock media Messages

Analogous to block DTMF and unblock DTMF but blocks media packets instead of DTMF packets. DTMF packets can still pass through when media blocking is enabled. Media packets can be blocked for an entire call, or directionally for individual participants. See block DTMF above for details.

In addition to blocking media for just one call participant, it's possible to block media for just a single media flow. This is relevant to scenarios that involve forked media that were established with one or more subscribe request. To select just one media flow for media blocking, in addition to selecting a source call participant as above, a destination call participant must be specified using the to-tag or to-labelkey in the message.

Another possibility to block media for individual media flows is to use one of the special all= keywords instead of directly specifying a single to-tag or to-label. With all=offer-answer all media flows from the given from-tag that resulted from an offer/answer negotiation are affected. Respectively with all=except-offer-answer the opposite happens. With all=flows all currently established media flows are affected regardless or how they were created.

silence media and unsilence media Messages

Identical to block media and unblock media except that media packets are not simply blocked, but rather have their payload replaced with silence audio. This is only supported for certain trivial audio codecs (i.e. G.711, G.722).

start forwarding and stop forwarding Messages

These messages control the recording daemon's mechanism to forward PCM via TCP/TLS. Unlike the call recording mechanism, forwarding can be enabled for individual participants (directionally) only, therefore these messages can be used with the same options as the block and unblock messages above. The PCM forwarding mechanism is independent of the call recording mechanism, and so forwarding and recording can be started and stopped independently of each other.

play media Message

Only available if compiled with transcoding support. The message must contain the key call-id and one of the participant selection keys described under the block DTMF message (such as from-tag, address, or label). Alternatively, the all flag can be set to play the media to all involved call parties.

Starts playback of a provided media file to the selected call participant. The format of the media file can be anything that is supported by ffmpeg, for example a .wav or .mp3 file. It will automatically be resampled and transcoded to the appropriate sampling rate and codec. The selected participant's first listed (preferred) codec that is supported will be chosen for this purpose.

Media files can be provided through one of these keys:

  • file

    Contains a string that points to a file on the local file system. File names can be relative to the daemon's working direction.

  • blob

    Contains a binary blob (string) of the contents of a media file. Due to the limitations of the ng transport protocol, only very short files can be provided this way, and so this is primarily useful for testing and debugging.

  • db-id

    Contains an integer. This requires the daemon to be configured for accessing a MySQL (or MariaDB) database via (at the minimum) the mysql-host and mysql-query config keys. The daemon will then retrieve the media file as a binary blob (not a file name!) from the database via the provided query.

  • repeat-times

    Contains an integer. How many times to repeat playback of the media. Default is 1.

  • start-pos

    Contains an integer. The start frame position to begin the playback from.

In addition to the result key, the response dictionary may contain the key duration if the length of the media file could be determined. The duration is given as in integer representing milliseconds.

stop media Message

Stops the playback previously started by a play media message. Media playback stops automatically when the end of the media file is reached, so this message is only useful for prematurely stopping playback. The same participant selection keys as for the play media message can and must be used. Will return the last frame played in last-frame-pos key.

play DTMF Message

Instructs rtpengine to inject a DTMF tone or event into a running audio stream. A call participant must be selected in the same way as described under the play media message above (including the possibility of using the all flag). The selected call participant is the one generating the DTMF event, not the one receiving it.

The dictionary key code (or alternatively digit) must be present in the message, indicating the DTMF event to be generated. It can be either an integer with values 0-15, or a string containing a single character (0 - 9, *, #, A - D). Additional optional dictionary keys are: duration indicating the duration of the event in milliseconds (defaults to 250 ms, with a minimum of 100 and a maximum of 5000); volume indicating the volume in absolute decibels (defaults to -8 dB, with 0 being the maximum volume and positive integers being interpreted as negative); and pause indicating the pause in between consecutive DTMF events in milliseconds (defaults to 100 ms, with a minimum of 100 and a maximum of 5000).

This message can be used to implement application/dtmf-relay or application/dtmf payloads carried in SIP INFO messages. Multiple DTMF events can be queued up by issuing multiple consecutive play DTMF messages.

If the destination participant supports the telephone-event RTP payload type, then it will be used to send the DTMF event. Otherwise a PCM DTMF tone will be inserted into the audio stream. Audio samples received during a generated DTMF event will be suppressed.

The call must be marked for DTMF injection using the inject DTMF flag used in both offer and answer messages. Enabling this flag forces all audio to go through the transcoding engine, even if input and output codecs are the same (similar to DTMF transcoding, see above).

statistics Message

Returns a set of general statistics metrics with identical content and format as the list jsonstats CLI command. Sample return dictionary:

  "statistics": {
    "currentstatistics": {
      "sessionsown": 0,
      "sessionsforeign": 0,
      "sessionstotal": 0,
      "transcodedmedia": 0,
      "packetrate": 0,
      "byterate": 0,
      "errorrate": 0
    "totalstatistics": {
      "uptime": "18",
      "managedsessions": 0,
      "rejectedsessions": 0,
      "timeoutsessions": 0,
      "silenttimeoutsessions": 0,
      "finaltimeoutsessions": 0,
      "offertimeoutsessions": 0,
      "regularterminatedsessions": 0,
      "forcedterminatedsessions": 0,
      "relayedpackets": 0,
      "relayedpacketerrors": 0,
      "zerowaystreams": 0,
      "onewaystreams": 0,
      "avgcallduration": "0.000000"
    "intervalstatistics": {
      "totalcallsduration": "0.000000",
      "minmanagedsessions": 0,
      "maxmanagedsessions": 0,
      "minofferdelay": "0.000000",
      "maxofferdelay": "0.000000",
      "avgofferdelay": "0.000000",
      "minanswerdelay": "0.000000",
      "maxanswerdelay": "0.000000",
      "avganswerdelay": "0.000000",
      "mindeletedelay": "0.000000",
      "maxdeletedelay": "0.000000",
      "avgdeletedelay": "0.000000",
      "minofferrequestrate": 0,
      "maxofferrequestrate": 0,
      "avgofferrequestrate": 0,
      "minanswerrequestrate": 0,
      "maxanswerrequestrate": 0,
      "avganswerrequestrate": 0,
      "mindeleterequestrate": 0,
      "maxdeleterequestrate": 0,
      "avgdeleterequestrate": 0
    "controlstatistics": {
      "proxies": [
	  "proxy": "",
	  "pingcount": 0,
	  "offercount": 0,
	  "answercount": 0,
	  "deletecount": 0,
	  "querycount": 0,
	  "listcount": 0,
	  "startreccount": 0,
	  "stopreccount": 0,
	  "startfwdcount": 0,
	  "stopfwdcount": 0,
	  "blkdtmfcount": 0,
	  "unblkdtmfcount": 0,
	  "blkmedia": 0,
	  "unblkmedia": 0,
	  "playmedia": 0,
	  "stopmedia": 0,
	  "playdtmf": 0,
	  "statistics": 0,
	  "errorcount": 0
      "totalpingcount": 0,
      "totaloffercount": 0,
      "totalanswercount": 0,
      "totaldeletecount": 0,
      "totalquerycount": 0,
      "totallistcount": 0,
      "totalstartreccount": 0,
      "totalstopreccount": 0,
      "totalstartfwdcount": 0,
      "totalstopfwdcount": 0,
      "totalblkdtmfcount": 0,
      "totalunblkdtmfcount": 0,
      "totalblkmedia": 0,
      "totalunblkmedia": 0,
      "totalplaymedia": 0,
      "totalstopmedia": 0,
      "totalplaydtmf": 0,
      "totalstatistics": 0,
      "totalerrorcount": 0
  "result": "ok"

publish Message

Similar to an offer message except that it is used outside of an offer/answer scenario. The media described by the SDP is published to rtpengine directly, and other peer can then subscribe to the published media to receive a copy.

The message must include the key sdp which should describe sendonly media; and the key call-id and from-tag to identify the publisher. Most other keys and options supported by offer are also supported for publish.

The reply message will contain an answer SDP in sdp, but unlike with offer this is not a rewritten version of the received SDP, but rather a recvonly answer SDP generated by rtpengine locally. Only one codec for each media section will be listed, and by default this will be the first supported codec from the published media. This can be influenced with the codec options described above, in particular the accept option.

The list of codecs given in the accept option is treated as a list of codec preferences, with the first codec listed being the most preferred codec to be accepted, and so on. It is allowable to list codecs that are not supported for transcoding. If no codecs from the accept list are present in the offer, then the first codec that is supported for transcoding is selected. If no such codec is present, then the offer is rejected. The special string any can be given in the accept list to influence this behaviour: If any is listed, then the first codec from the offer is accepted even if it's not supported for transcoding.

subscribe request Message

This message is used to request subscription (i.e. receiving a copy of the media) to one or multiple existing call participants, which must have been created either through the offer/answer mechanism, or through the publish mechanism.

A single call participant can be selected in the same way as described under block DTMF. Multiple call participants can be selected either by using the all keyword, in which case all call participants that were created through the offer/answer mechanism will be selected, or by providing a list of tags (from-tags) in the from-tags list.

This message then creates a new call participant, which corresponds to the subscription. This new call participant will be identified by a newly generated unique tag, or by the tag given in the to-tag key. If a label is to be set for the newly created subscription, it can be set through set-label.

The reply message will contain a sendonly offer SDP in sdp which by default will mirror the SDP of the call participant being subscribed to. If multiple call participants are subscribed to at the same time, then this SDP will contain multiple media sections, combined out of the media sections of all selected call participants. This offer SDP can be manipulated with the same flags as used in an offer message, including the option to manipulate the codecs. The reply message will also contain the from-tags (corresponding to the call participants being subscribed to) and the to-tag (corresponding to the subscription, either generated or taken from the received message).

If a subscribe request is made for an existing to-tag then all existing subscriptions for that to-tag are deleted before the new subscriptions are created.

subscribe answer Message

This message is expected to be received after responding to a subscribe request message. The message should contain the same to-tag as the reply to the subscribe request as well as the answer SDP in sdp.

By default, the answer SDP must accept all codecs that were presented in the offer SDP (given in the reply to subscribe request). If not all codecs were accepted, then the subscribe answer will be rejected. This behaviour can be changed by including the allow transcoding flag in the message. If this flag is present, then the answer SDP will be accepted as long as at least one valid codec is present, and the media will be transcoded as required. This also holds true if some codecs were added for transcoding in the subscribe request message, which means that allow transcoding must always be included in subscribe answer if any transcoding is to be allowed.

The reply message will simply indicate success or failure. If successful, media forwarding will start to the endpoint given in the answer SDP.

unsubscribe Message

This message is a counterpart to subsscribe answer to stop an established subscription. The subscription to be stopped is identified by the to-tag.

The tcp-ng Control Protocol

rtpengine also has support for ng control protocol where transport is TCP (If enabled in the config via the --listen-tcp-ng option). Everything said for UDP based ng protocol counts for TCP variant too.

HTTP/WebSocket support

If enabled in the config, rtpengine can handle requests made to it via HTTP, HTTPS, or WebSocket (WS or WSS) connections. The supported HTTP URIs and WebSocket subprotocols are described below.

Dummy Test Interfaces

For HTTP and HTTPS, the URI /ping is provided, which simply responds with pong if requested via GET. For WebSockets, the subprotocol is provided, which simply echoes back any messages that are sent to it.

CLI Interface

This interface supports the same commands as the CLI tool rtpengine-ctl that comes packaged with rtpengine. For HTTP and HTTPS, the command is appended to the URI base /cli/ and the request is made via GET, with spaces replaced by plus signs as required by HTTP (e.g. GET /cli/list+totals). For WebSockets, the subprotocol is and each WebSocket message corresponds to one CLI command and produces one message in response. The format of each response is exactly the same as produced by the CLI tool rtpengine-ctl and therefore meant for plain text representation.

ng Protocol Interface

This interface can be used to send and receive ng protocol messages over HTTP or WebSocket connections instead of plain UDP.

For HTTP and HTTPS, the URI /ng is used, with the request being made by POST and the content-type set to application/x-rtpengine-ng. The message body must be in the same format as the body of an UDP-based ng message and must therefore consist of a unique cookie string, followed by a single space, followed by the message in bencode format. Likewise, the response will be in the same format, including the unique cookie.

For WebSockets, the subprotocol is used and the protocol follows the same format. Messages must consist of a unique cookie and a string in bencode format, and responses will also be in the same format.

Prometheus Stats Exporter

The Prometheus metrics can be found under the URI /metrics.

Janus Interface and Replacement Functionality

Rtpengine supports a limited and narrow subset of the features provided by Janus, specifically the basic business logic behind the videoroom plugin. This makes it possible to use rtpengine as a drop-in replacement for Janus for this one specific use case, and has the benefit of being able to use all the extra features that rtpengine provides, such as transcoding, in-kernel packet forwarding for improved performance, etc.

The required subset of the Janus API is exposed via rtpengine's HTTP/WS interface. The HTTP admin API is connected to the /admin URI path using a JSON payload (same as Janus does), while the module communication happens on the WS protocol janus-protocol, also with JSON payloads (same as Janus does). Unlike Janus, both HTTP and WS endpoints are running on the same port. In fact, there is no real distinction between both interfaces, therefore both admin and non-admin messages can be sent via either interface. HTTPS and WSS are also supported.

Token-based plugin authentication works similar to how it works in Janus except that only the single videoroom plugin is supported. The configuration setting janus-secret must be set to enable clients to connect to this simulated Janus interface and make use of its features.

Under the hood the functionality of the videoroom plugin is facilitated using rtpengine's publish and subscribe methods, which are mapped directly to the respective Janus methods. One Janus video room becomes one rtpengine call, with a distinctive and unique call ID based on the video room ID.

There's currently no support for customising the SDP features and options used within the Janus drop-in mode, and, as Janus is WebRTC-specific, all SDPs produced from this mode can be used directly by WebRTC clients. Non-WebRTC clients can participate in the same video room as Janus clients if the respective mapped publish and subscribe methods are used, and with the call ID mapped to the video room ID.