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API description

The API is pretty minimalistic. It is limited to a single call at a time and provides the following functionality:

  • Register to the PBX
  • Make a call or accept a call
  • Send audio files
  • Receive DTMF tones
  • Hang up

This is all I needed and I guess it brings ToT devices a big step ahead (and also helps the beginner to softly make his feet wet with the complex world of SIP, SDP, and RDP).

Flexosip uses a minimalistic approach on callback functions. Similar to inheritance in object-oriented languages, flexosip already provides an implementation of all the callbacks. However, they can be overridden in your application by just defining them again with the same name.

(There is no magic to this, but just the power of Weak symbols. And no, it does not allow real subclassing or multiple inheritance or anything, just overriding the default implementation once.)


#include "flexosip/flexosip.h"
#include "flexosip/flexosnd.h"
#include "flexosip/flexortp.h"

To initialize debugging, call TRACE_INITIALIZE(OSIP_INFO1, NULL);, if you want to. (Instead of OSIP_INFO1, you can also use another verbosity level.)

If you want some cleanup to happen when you Ctrl-C the program, add the following:

signal(SIGINT, fesip_cleanup);
signal(SIGHUP, fesip_cleanup);
signal(SIGTERM, fesip_cleanup);
signal(SIGQUIT, fesip_cleanup);

And then, initialize the network subsystem:

fesip_listen(IPPROTO_UDP, false, 0);

Other options are described in flexosip.h.


Before you can make or receive a call, you need to register with your PBX (aka your telephone switch or VoIP exchange). The PBX is know as the registrar in SIP parlance.

You do this by calling

fesip_register(uri, registrar, login, password);

where the uri is typically of the form sip:<extension>@<registrar> and the registrar sip:<hostname or domain>, e.g. and, respectively.

The SIP registration (like almost all SIP messages) happen in the background while your code continues executing. If you want to be sure that the registration was successful (i.e., correct credentials were supplied) or need the successful registration to continue (namely, making an outgoing call), you can do

ok = fesip_wait_registered();

which will wait and return < 0 for failure. This will wait up to about 10 seconds or for two registration failures. (One registration failure is normal, as the first registration attempt is done without credentials.)

Now, your device is registered and can send and receive calls.

Event loop

In your main code, you will need an event loop as follows:

while (1) {
  // Handle SIP and RTP (audio)

  // Do your own stuff

fesip_handle_event() will sleep between 10 and 20 ms, depending on what it has to do.

The rest of your processing loop should not take more than 10 ms, unless you know that no audio is currently playing.

Receive calls

When an incoming call arrives, the event handler will call your fesip_event_invite() callback:

int fesip_event_invite(eXosip_event_t *evt,
    const char *host, int port, int format)

This is passed the incoming call event, the host and port that audio should go to, and the expected audio format (currently only PCMA/8000, aka 8 kHz A-Law). For now, you probably want to ignore them all.

To decline the call, just return SIP_BUSY_HERE.

However, if the call should be accepted, that function should return SIP_RINGING for now and set a variable, which will trigger accepting the connection in the event loop. This is necessary due to the locking that happens in fesip_handle_event().

In general, you cannot call any `fesip_*()` library functions from 
any of the event handlers (`fesip_event_*()`). The only exception 
currently is `fesip_play()`.

Outside the event handler, back in the event loop, you call


to accept the call. If you want the caller to hear some ringing first, feel free to do this only after a few seconds.

Play audio

You can then send an audio message with


where filename is any 16 kHz mono file understood by libsndfile. I use .ogg files.

fesip_play() actually adds the file to the end of the play queue, so you can call it multiple times. To stop playing and clear the play queue (e.g., on an incoming DTMF event), call


(notice the fesnd_* prefix).

Make calls

To make a call, use

fesip_call(from, to, subject, NULL);

where from is your own SIP address, to is the destination's, and subject is the subject trasmitted (may or may not be displayed).

When the remote side declines, your

void fesip_event_terminate(eXosip_event_t *evt);

Callback will be called (which you do not need to implement, if you do not care). This you will get on all call terminations, incoming or outgoing, pending or active.

When it answers, however, the callback will be

void fesip_event_answered(eXosip_event_t *evt);

There, you can decline the call with fesip_terminate_nolock() (created especially to be called from a callback) or start playing audio as explained above.

Receive DTMF

When the remote side — in an incoming or outgoing call — presses a key,

void fesip_event_dtmf(char c);

is called with the ASCII character corresponding to the key pressed (typically, '0'…'9', '*', '#').

The end

That is already everything you need to know. Now you can start your own ToT device!

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