Provided by: libexosip2-dev_4.0.0-4ubuntu1_amd64 
NAME
How-To initiate, modify or terminate calls. - eXosip2 offers a flexible API to help you
controling calls.
Initiate a call
To start an outgoing call, you typically need a few headers which will be used by eXosip2
to build a default SIP INVITE request. The code below is used to start a call:
osip_message_t *invite;
int cid;
int i;
i = eXosip_call_build_initial_invite (ctx, &invite, "<sip:to@antisip.com>",
"<sip:from@antisip.com>",
NULL, // optional route header
"This is a call for a conversation");
if (i != 0)
{
return -1;
}
osip_message_set_supported (invite, "100rel");
{
char tmp[4096];
char localip[128];
eXosip_guess_localip (ctx, AF_INET, localip, 128);
snprintf (tmp,04096,
"v=0 0
"o=jack 0 0 IN 0P4 %s
"s=conversat0on
"c=IN 0P4 %s
"t=0 0 0
"m=audio %s RTP/AVP 008 101
"a=rtpmap:0 PCMU/80000
"a=rtpmap:8 PCMA/8000 0
"a=rtpmap:101 te0,plocalip,nlocalip, port);
"a=fmtp:101 0-11
osip_message_set_body (invite, tmp, strlen (tmp));
osip_message_set_content_type (invite, "application/sdp");
}
eXosip_lock (ctx);
cid = eXosip_call_send_initial_invite (ctx, invite);
if (cid > 0)
{
eXosip_call_set_reference (ctx, i, reference);
}
eXosip_unlock (ctx);
return i;
The above code is using eXosip_call_build_initial_invite to build a default SIP INVITE
request for a new call. You have to insert a SDP body announcing your audio parameter for
the RTP stream.
The above code also show the flexibility of the eXosip2 API which allow you to insert
additionnal headers such as 'Supported: 100rel' (announcing support for a SIP extension).
Thus you can enterely control the creation of SIP requests.
The returned element of eXosip_call_send_initial_invite is the cid (call identifier) that
you can use to send a CANCEL. In future events other than 100 Trying, you'll also get the
did (dialog identifier) that will also be needed to control established calls.
eXosip_call_set_reference is also a mean to attach one of your own context to a call so
that you'll get your pointer back in eXosip_event.
Answer a call
The code below is another example that teach you how to answer an incoming call.
You'll usually need to send a '180 Ringing' SIP answer when receiving a SIP INVITE:
eXosip_lock (ctx);
eXosip_call_send_answer (ctx, evt->tid, 180, NULL);
eXosip_unlock (ctx);
Note: The above code also shows that the stack is sometimes able to build and send a
default SIP messages with only one API call
Then, when the user wants to answer the call, you'll need to send a 200 ok and insert a
SDP body in your SIP answer:
osip_message_t *answer = NULL;
eXosip_lock (ctx);
i = eXosip_call_build_answer (ctx, evt->tid, 200, &answer);
if (i != 0)
{
eXosip_call_send_answer (ctx, evt->tid, 400, NULL);
}
else
{
i = sdp_complete_200ok (evt->did, answer);
if (i != 0)
{
osip_message_free (answer);
eXosip_call_send_answer (ctx, evt->tid, 415, NULL);
}
else
eXosip_call_send_answer (ctx, evt->tid, 200, answer);
}
eXosip_unlock (ctx);
Note: In the above code, you can note that to send a response to a request, you have to
use the tid (transaction identifier) and not a cid (call identifier) or a did (dialog
identifier).
Note2: For sending a 200ok, you'll usually need to insert a SDP body in the answer and
before this, to negotiate the parameters and codecs that you want to support. This is left
to you! Once you have created the SDP, you add it in the answer using the following code:
osip_message_set_body (answer, tmp, strlen (tmp));
osip_message_set_content_type (answer, "application/sdp");
Terminate a Call
Simple API, no much to say about it! You can use it when you want: it will either send a
CANCEL, a negative answer or a BYE depending on the call state.
eXosip_lock (ctx);
eXosip_call_terminate (ctx, cid, did);
eXosip_unlock (ctx);
Note: You can't stop a call where no 100 Trying has been received. In that case, you need
to wait before sending a CANCEL or a BYE... This is per rfc3261.
Sending INFO, REFER, UPDATE, NOTIFY, OPTIONS request
The call control API allows you to send and receive REFER, UPDATE, INFO, OPTIONS, NOTIFY
and INVITEs whitin calls.
Here you have a code sample to send an INFO requests used to send an out of band dtmf
within the signalling layer. (not standard, but still used on some system!)
osip_message_t *info;
char dtmf_body[1000];
int i;
eXosip_lock (ctx);
i = eXosip_call_build_info (ctx, evt->did, &info);
if (i == 0) 0, c);
{ 0uration=250
snprintf (dtmf_body, 999, "Signal=%c
osip_message_set_content_type (info, "application/dtmf-relay");
osip_message_set_body (info, dtmf_body, strlen (dtmf_body));
i = eXosip_call_send_request (ctx, evt->did, info);
}
eXosip_unlock (ctx);
Sending any other request, with any header
You can in fact, send any kind of other request using eXosip2 API.
You will find many other API to build any kind of sip message. Using osip API, you can add
any header or body in those message. eXosip2 will always prepare the minimal and technical
stuff you need.
osip_message_t *message;
char body[1000];
int i;
eXosip_lock (ctx);
i = eXosip_call_build_request (ctx, evt->did, "PRIVATECOMMAND", &message);
if (i == 0) 0);
{ 0oom=2;light=off
snprintf (body, 999, "room=1;light=on
osip_message_set_content_type (message, "application/antisip-domotic");
osip_message_set_body (message, body, strlen (body));
osip_message_set_header (invite, "P-MyCommand", "option=value");
i = eXosip_call_send_request (ctx, evt->did, message);
}
eXosip_unlock (ctx);
Author
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Version 4.0.0 Fri Jan 17How-To initiate, modify or terminate calls.(3)