[Asterisk-video] capability translation from h245 to SIP
Sergio Garcia Murillo
sergio.garcia at fontventa.com
Thu Sep 11 02:03:57 CDT 2008
I wouldn't waste much effort supporting MPEG4, the improvement over h263
is very low and not many softphones support it (although most network
cameras do, but don't know if they would conform the bitrate as needed).
The only important issue to support it is that currently only one codec
is supported for audio and video, so in order to support multiple ones
we'll have to add some configuration logic to the library to choose the
best codec available.
Until h264 phones are more widely available I wouldn't spend much time
on it.
Best regards
Sergio
Dan Julius escribió:
> Most of the phones I've tested with also support MPEG4
>
> |receiveVideoCapability genericVideoCapability standard 0.0.8.245.1.0.0
>
> maxBitRate = 521|
>
>
> I think it would make much to support this codec as well.
> Can you give me pointers as to which parts of the code you think might
> need to be modified?
>
> Thanks,
> Dan
>
> On Thu, Sep 11, 2008 at 1:02 AM, Sergio Garcia Murillo
> <sergio.garcia at fontventa.com <mailto:sergio.garcia at fontventa.com>> wrote:
>
> 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
>>>
>>>
>>> ------------------------------------------------------------------------
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
>>> asterisk-video mailing list
>>> To UNSUBSCRIBE or update options visit:
>>> http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by
>> http://www.api-digital.com--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-video
>>
>>
>> ------------------------------------------------------------------------
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> asterisk-video mailing list
>> To UNSUBSCRIBE or update options visit:
>> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> asterisk-video mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-video
-------------- next part --------------
An HTML attachment was scrubbed...
URL: http://lists.digium.com/pipermail/asterisk-video/attachments/20080911/2d771e80/attachment-0001.htm
More information about the asterisk-video
mailing list