A RTP stack for Go
Go
Switch branches/tags
Nothing to show
Clone or download
Latest commit 324aada Jun 11, 2018

README.md

RTP/RTCP stack for Go

This Go package implements a RTP/RTCP stack for Go. The package is a sub-package of the standard Go net package and uses standard net package functions.

How to build

The rtp sources use the GOPATH directory structure. To build, test, and run the software just add the main goRTP directory to GOPATH. For further information about this structure run go help gopath and follow the instructions. The rtp package is below the package net to make clear that rtp is a network related package.

To build the package just run go build net/rtp and then go install net/rtp. To excecute the tests just run go test net/rtp. The tests check if the code works with the current Go installation on your system. It should PASS.

A demo program is available and is called rtpmain. Use go build net/rtpmain to build it. The command go install net/rtpmain installs it in the bin directory of the main directory.

How to use

This is a pure RTP / RTCP stack and it does not contain any media processing, for example generating or packing the payload for audio or video codecs.

The directory src/net/rtpmain contains an example Go program that performs a RTP some tests on localhost that shows how to setup a RTP session, an output stream and how to send and receive RTP data and control events. Parts of this program are used in the package documentation.

The software should be ready to use for many RTP applications. Standard point-to-point RTP applications should not pose any problems. RTP multi-cast using IP multi-cast addresses is not supported. If somebody really requires IP multi-cast it could be added at the transport level.

RTCP reporting works without support from application. The stack reports RTCP packets and if the stack created new input streams and an application may connect to the control channel to receive the RTCP events. Just have a look into the example program. The RTCP fields in the stream structures are accessible - however, to use them you may need to have some know-how of the RTCP definitions and reporting.

The documentation

After you downloaded the code you may use standard godoc to get a nice formatted documentation. Just change into the src directory, run godoc -http=:6060 -path=".", point your browser at localhost:6060, and select src at the top of the page.

I've added some package global documentation and tried to document the globally visible methods and functions.

Before you start hacking please have a look into the documentation first, in particular the package documentation (doc.go).

Some noteable features

  • The current release V1.0.0 computes the RTCP intervals based on the length of RTCP compound packets and the bandwidth allocated to RTCP. The application may set the bandwidth, if no set GoRTP makes somes educated guesses.

  • The application may set the maximum number of output and input streams even while the RTP session is active. If the application des not set GoRTP sets the values to 5 and 30 respectively.

  • GoRTP produces SR and RR reports and the associated SDES for active streams only, thus it implements the activity check as defined in chapter 6.4

  • An appplication may use GoRTP in simple RTP mode. In this mode only RTP data packets are exchanged between the peers. No RTCP service is active, no statistic counters, and GoRTP discards RTCP packets it receives.

  • GoRTP limits the number of RR to 31 per RTCP report interval. GoRTP does not add an additional RR packet in case it detects more than 31 active input streams. This restriction is mainly due to MTU contraints of modern Ethernet or DSL based networks. The MTU is usually about 1500 bytes, GoRTP limits the RTP/RTCP packet size to 1200 bytes. The length of an RR is 24 bytes, thus 31 RR already require 774 bytes. Adding some data for SR and SDES fills the rest.

  • An application may register to a control event channel and GoRTP delivers a nice set of control and error events. The events cover:

    • Creation of a new input stream when receiving an RTP or RTCP packet and the SSRC was not known
    • RTCP events to inform about RTCP packets and received reports
    • Error events
  • Currently GoRTP supports only SR, RR, SDES, and BYE RTCP packets. Inside SDES GoRTP does not support SDES Private and SDES H.323 items.