[Asterisk-video] capability translation from h245 to SIP
Sergio Garcia Murillo
sergio.garcia at fontventa.com
Wed Sep 10 17:02:49 CDT 2008
In Asterisk 1.4 the only thing that can be configured for video is de
codec, so there is not much problem as the only one supported is
h263-1998/2000 for sending and h263-1996 h263-1998/2000 for receiving.
Using the videocaps branch we could map the codec parameters to the SDP
offer.
Best regards
Sergio
Dan Julius escribió:
> Hi, Sergio,
>
> If there is no capability mapping, how does asterisk decide what
> capabilities to put in the SDP body of the SIP Invite message?
>
>
> When I originate a call from a Nokia N73 to a SIP client, the Invite
> message includes and SDP section for video:
> m=video 10070 RTP/AVP 34 103
> a=rtpmap:103 h263-1998/90000
> a=sendrecv
>
>
> When I originate a call from a Nokia N95 the SDP is different and also
> includes the h263/9000 capability:
> m=video 10120 RTP/AVP 34 103
>
> a=rtpmap:34 H263/90000
> a=rtpmap:103 h263-1998/90000
> a=sendrecv
>
>
>
>
> Thanks,
> Dan
>
> On Tue, Sep 9, 2008 at 11:54 AM, Sergio Garcia Murillo
> <sergio.garcia at fontventa.com <mailto:sergio.garcia at fontventa.com>> wrote:
>
> Hi Dan,
>
> There is no capability negotiation, or mapping between SIP and the
> h245, I send fixed ones For H263 and AMR:
>
>
> [0]={
> capabilityTableEntryNumber = 1
> capability = receiveAndTransmitVideoCapability
> h263VideoCapability {
> qcifMPI = 2
> maxBitRate = 520
> unrestrictedVector = FALSE
> arithmeticCoding = FALSE
> advancedPrediction = FALSE
> pbFrames = FALSE
> temporalSpatialTradeOffCapability = FALSE
> errorCompensation = FALSE
> }
> }
> [1]={
> capabilityTableEntryNumber = 2
> capability = receiveAndTransmitAudioCapability
> genericAudioCapability {
> capabilityIdentifier = standard 0.0.8.245.1.1.1
> maxBitRate = 122
> collapsing = 1 entries {
> [0]={
> parameterIdentifier = standard 0
> parameterValue = unsignedMin 1
> }
> }
> }
> }
> .
>
> Best regards
> Sergio
>
> Dan Julius escribió:
>> Hi,
>>
>> I'm experiencing some issues when using Asterisk as a gateway
>> between 3G and SIP clients.
>> I'm trying to complete a call using h263p as the video codec on
>> the sip client in order to avoid video transcoding.
>> For audio I'm using AMR on the 3G side and g711 on the SIP side
>> since most SIP clients don't support AMR out of the box.
>>
>> For some phones this works without a problem while other phones
>> don't seem to show video depending on the SIP client.
>> For example: N73 works with X-Lite, but not with OpenPhone. N95
>> works with both.
>> I've traced this down to an issue with the capability exchange,
>> specifically how capabilities are translated from h245 to SIP.
>>
>> Can someone point to which part of the code, Asterisk or libh324M
>> does the mapping between capabilities?
>> After reviewing some relevant RFCs and specs, it is not clear to
>> me what the 1:1 mapping should be - is there any relevant
>> documentation?
>>
>> Thanks,
>> Dan
>>
>>
>> ------------------------------------------------------------------------
>>
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>
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