<div>Hi,<br clear="all"></div><div><br></div><div>I'm using SIPp program to make a call toward Asterisk. The call opens a channel that support video with the following SDP</div><div><br></div><div>INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0<br>
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]<br> From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]<br> To: sut <sip:[service]@[remote_ip]:[remote_port]><br> Call-ID: [call_id]<br>
CSeq: 1 INVITE<br> Contact: sip:sipp@[local_ip]:[local_port]<br> Max-Forwards: 70<br> Subject: Dummy User<br> User-Agent: sipp<br> Content-Type: application/sdp<br> Content-Length: [len]<br>
<br> v=0<br> o=sipp 53655765 2353687637 IN IP[local_ip_type] [local_ip]<br> s=-<br> c=IN IP[local_ip_type] [local_ip]<br> t=0 0<br> <br> <strong>m=audio [auto_media_port] RTP/AVP 0 8 101 <br>
a=fmtp:101 0-16 <br> a=rtpmap:0 PCMU/8000<br> a=rtpmap:8 PCMA/8000<br> a=rtpmap:101 telephone-event/8000 <br> a=sendrecv <br> m=video [media_port] RTP/AVP 115<br> a=rtpmap:115 H263-1998/90000<br>
a=sendrecv</strong><br></div><div><br></div><div>Now where can I find the audio/video port in ast_channel?</div><div><br></div><div>Thank in advance</div><br>-- <br>_______________________________________<br>Salvatore Frandina<br>
website: <a href="http://frandinas.altervista.org" target="_blank">http://frandinas.altervista.org</a><br>mail: <a href="mailto:salvatore.frandina@gmail.com" target="_blank">salvatore.frandina@gmail.com</a><br><br>_______________________________________<br>
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