On Thu, Jun 13, 2013 at 12:04 PM, <[email protected]> wrote: > Hi there > > I have asterisk 10.11.1 which seems to have problem negotiating codec. > > Scenario: SIP PHONE1 (XLite) extension 1003, allowed codecs alaw, h263p > and SIP phone2 (Grandstream GXV3175) extension 1004, allowed codec alaw, > h263p. I have tried similar combination of codecs and SIP phone but when > making a video call, it report "Peer doesn't provide video". It seems > Asterisk is failing to set capability correct. Both codecs are enabled on > the SIP Phones > > <snip>
The 200 OK response from the called XLite phone is declining the video stream: <--- SIP read from UDP:10.10.10.129:48464 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.10.10.105:5060;branch=z9hG4bK368135b0;rport=5060 Contact: <sip:[email protected]:48464> To: "SAM"<sip:[email protected]>;tag=0c90cc0c From: <sip:[email protected]>;tag=as24914503 Call-ID: MmNjOTczNDU5YjZmYjAyNWMxY2Q1MDZjODdhYzQwZjA CSeq: 102 INVITE Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Supported: replaces, eventlist User-Agent: X-Lite release 4.5.2 stamp 70142 Content-Length: 234 v=0 o=- 13015615910543193 2 IN IP4 10.10.10.129 s=X-Lite 4 release 4.5.2 stamp 70142 c=IN IP4 10.10.10.129 t=0 0 m=audio 53188 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv m=video 0 RTP/AVP 115 <-------------> --- (12 headers 10 lines) --- Found RTP audio format 8 Found RTP audio format 101 Found audio description format telephone-event for ID 101 Capabilities: us - (alaw|h263p), peer - audio=(alaw)/video=(nothing)/text=(nothing), combined - (alaw) Note that the port for the video stream is set to 0. Asterisk is doing the correct thing: it notes that the answer to its offer declined the video stream, so it disables video for the call between the two endpoints. Matt -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org
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