[asterisk-users] Codec Negotiation problem

[research at businesstz.com]

Hi Matt

Thanks for your response. I have tried with two GXV3175 with same result.
Let me dig deep on this to find out the route cause

Sam
Matthew Jordan wrote:
> On Thu, Jun 13, 2013 at 12:04 PM, <research at businesstz.com> 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:1003 at 10.10.10.129:48464>
> To: "SAM"<sip:1003 at 10.10.10.105>;tag=0c90cc0c
> From: <sip:1004 at 10.10.10.105>;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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