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<title>libeXosip2: How-To initiate, modify or terminate calls.</title>
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<h1>How-To initiate, modify or terminate calls.<br>
<small>
[<a class="el" href="group__libeXosip2.html">The eXtented eXosip stack</a>]</small>
</h1><table border="0" cellpadding="0" cellspacing="0">
<tr><td></td></tr>
</table>
eXosip2 offers a flexible API to help you controling calls.<p>
<h2>Initiate a call</h2>
<p>
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:<p>
<pre>
osip_message_t *invite;
int i;</pre><p>
<pre> i = eXosip_call_build_initial_invite (&invite, "<sip:to@antisip.com>",
"<sip:from@antisip.com>",
NULL, // optionnal route header
"This is a call for a conversation");
if (i != 0)
{
return -1;
}</pre><p>
<pre> osip_message_set_supported (invite, "100rel");</pre><p>
<pre> {
char tmp[4096];
char localip[128];</pre><p>
<pre> eXosip_guess_localip (AF_INET, localip, 128);
snprintf (tmp, 4096,
"v=0\r\n"
"o=josua 0 0 IN IP4 %s\r\n"
"s=conversation\r\n"
"c=IN IP4 %s\r\n"
"t=0 0\r\n"
"m=audio %s RTP/AVP 0 8 101\r\n"
"a=rtpmap:0 PCMU/8000\r\n"
"a=rtpmap:8 PCMA/8000\r\n"
"a=rtpmap:101 telephone-event/8000\r\n"
"a=fmtp:101 0-11\r\n", localip, localip, port);
osip_message_set_body (invite, tmp, strlen (tmp));
osip_message_set_content_type (invite, "application/sdp");
}</pre><p>
<pre> eXosip_lock ();
i = eXosip_call_send_initial_invite (invite);
if (i > 0)
{
eXosip_call_set_reference (i, reference);
}
eXosip_unlock ();
return i;</pre><p>
<pre></pre><p>
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.<p>
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.<p>
The returned element of eXosip_call_send_initial_invite is the call identifier that you can use to send a CANCEL. In future events other than 100 Trying, you'll also get the dialog identifier that will also be needed to control established calls.<p>
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 <a class="el" href="structeXosip__event.html">eXosip_event</a>.<p>
<h2>Answer a call</h2>
<p>
The code below is another example that teach you how to answer an incoming call.<p>
You'll usually need to send a "180 Ringing" SIP answer when receiving a SIP INVITE:<p>
<pre>
eXosip_lock ();
eXosip_call_send_answer (ca->tid, 180, NULL);
eXosip_unlock ();
</pre><p>
<b>Note</b>: The above code also shows that the stack is sometimes able to build and send a default SIP messages with only one API call<p>
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:<p>
<pre>
osip_message_t *answer = NULL;</pre><p>
<pre> eXosip_lock ();
i = eXosip_call_build_answer (ca->tid, 200, &answer);
if (i != 0)
{
eXosip_call_send_answer (ca->tid, 400, NULL);
}
else
{
i = sdp_complete_200ok (ca->did, answer);
if (i != 0)
{
osip_message_free (answer);
eXosip_call_send_answer (ca->tid, 415, NULL);
}
else
eXosip_call_send_answer (ca->tid, 200, answer);
}
eXosip_unlock ();
</pre><p>
<b>Note</b>: In the above code, you can note that to send a response to a request, you have to use the transaction identifier (and not a call identifier or a dialog identifier!)<p>
<b>Note2</b>: 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. In the test tool, provided by eXosip2 (josua application), you'll find a very basic implementation of the SDP negotiation.<p>
<h2>Sending other request</h2>
<p>
The call control API allows you to send and receive REFER, UPDATE, INFO, OPTIONS, NOTIFY and INVITEs whitin calls. A few limitations still exist for answering other requests within calls, but it should be already possible to send any kind of request.<p>
Here you have a code sample to send an INFO requests used to send an out of band dtmf within the signalling layer.<p>
<pre>
osip_message_t *info;
char dtmf_body[1000];
int i;</pre><p>
<pre> eXosip_lock ();
i = eXosip_call_build_info (ca->did, &info);
if (i == 0)
{
snprintf (dtmf_body, 999, "Signal=%c\r\nDuration=250\r\n", 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 (ca->did, info);
}
eXosip_unlock ();
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