[asterisk-dev] Asterisk audio/video RTP question

[Salvatore Frandina]

Hi,

I'm using SIPp program to make a call toward Asterisk. The call opens a
channel that support video with the following SDP

INVITE sip:[service]@[remote_ip]:[remote_port] SIP/2.0
Via: SIP/2.0/[transport] [local_ip]:[local_port];branch=[branch]
From: sipp <sip:sipp@[local_ip]:[local_port]>;tag=[call_number]
To: sut <sip:[service]@[remote_ip]:[remote_port]>
Call-ID: [call_id]
CSeq: 1 INVITE
Contact: sip:sipp@[local_ip]:[local_port]
Max-Forwards: 70
Subject: Dummy User
User-Agent: sipp
Content-Type: application/sdp
Content-Length: [len]

v=0
o=sipp 53655765 2353687637 IN IP[local_ip_type] [local_ip]
s=-
c=IN IP[local_ip_type] [local_ip]
t=0 0

*m=audio [auto_media_port] RTP/AVP 0 8 101
a=fmtp:101 0-16
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=sendrecv
m=video [media_port] RTP/AVP 115
a=rtpmap:115 H263-1998/90000
a=sendrecv*

Now where can I find the audio/video port in ast_channel?

Thank in advance

-- 
_______________________________________
Salvatore Frandina
website: http://frandinas.altervista.org
mail: salvatore.frandina at gmail.com

_______________________________________
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