Код: Выделить всё
> <--- Reliably Transmitting (NAT) to 172.16.2.171:5060 --->
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP
> 192.168.200.5:5060;branch=z9hG4bK164066070;received=172.16.2.171;rport=5060
> From: <sip:103@172.16.2.171>;tag=810607892
> To: <sip:104@172.16.2.170>;tag=as0e6ebfc4
> Call-ID: 729679979
> CSeq: 21 INVITE
> User-Agent: Asterisk PBX 1.6.0.26-FONCORE-r78
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
> INFO
> Supported: replaces, timer
> Contact: <sip:103@172.16.2.171>
> Content-Type: application/sdp
> Content-Length: 403
>
> v=0
> o=root 1331097612 1331097613 IN IP4 172.16.2.171
> s=Asterisk PBX 1.6.0.26-FONCORE-r78
> c=IN IP4 172.16.2.171
> b=CT:384
> t=0 0
> m=audio 36674 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=silenceSupp:off - - - -
> a=ptime:20
> a=sendrecv
> m=video 36606 RTP/AVP 34 102
> a=rtpmap:34 H263/90000
> a=rtpmap:102 H264/90000
> a=sendrecv