One way audio with SILK

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Hi Alex,

Afaik, asymmetric PT is allowed by standard, but symmetric PT is
recommended, so there was ticket #1300
(https://trac.pjsip.org/repos/ticket/1300). Currently PT cannot be
changed on run time, but as SILK is a 'custom' codec, you can assign
the SILK PT in its pjmedia_codec_info to any value that is not used by
other codecs.

Just FYI, we've just integrated SILK to SVN trunk (based on CSipSimple
implementation), see https://trac.pjsip.org/repos/ticket/1586, as now
it seems that we are running out of PT allocation, we may prioritize
this ticket https://trac.pjsip.org/repos/ticket/1297.

BR,
nanang


On Fri, Sep 21, 2012 at 8:42 PM, Alex <alex.solis at telcentris.com> wrote:
> Nope, I am not using it at lower level.
>
> The SDP negotiation looks ok. From the logs during an incoming off-net call,
> SILK fails to decode with the following error:
> silk.c  Failed to decode silk frame (error -12)
> which is SKP_SILK_DEC_PAYLOAD_ERROR
>
> There seems to be a server side work around so we will go that way (which is
> probably allow asym payloads).
>
> Thanks a lot for you help!
> Alex.
>
> On Sep 21, 2012, at 6:34 AM, R?gis Montoya wrote:
>
> Hi,
>
> Weird, by default pjsua_media doesn't allow asym negotiation. Are you using
> pjsua or something at lower level (like directly pjmedia for example)?
> If lower level (pjmedia sdp neg), you should ensure your don't use sdp neg
> with allow_asym flag set to 0 (PJ_FALSE).
>
> However, if SDP negotiation is correct (and you can check that easily as you
> have tcpdumps) and remote side is well implemented it should not be a
> problem to have asym PT. As far as I understood having things without asym
> allowed is for compatibility purpose with servers that doesn't allow asym on
> their side.
> // disclamer : I'm new on that part and didn't fully read the rfc on sdp neg
> yet -- so maybe I'm wrong ;)
>
> I tell that because, not sure it's your root problem. If payload type
> incoming is wrong (misunderstood by pj) you have logs of pjmedia crying
> about invalid PT. If you don't see these logs, it's probably understanding
> correctly what is sent by the server, and decode it correctly and you should
> have incoming sound at least.
>
>
> 2012/9/21 Alex <alex.solis at telcentris.com>
>>
>> Hi Regis,
>>
>> Looks like the problem is that it is using 104 for decoding and 121 for
>> encoding. Is there a config option in pjsip to force the codec use the same
>> payload for both encoding and decoding?
>>
>> Thanks,
>> Alex.
>>
>> On Sep 19, 2012, at 11:56 PM, R?gis Montoya wrote:
>>
>> Hi Alex
>>
>> If you are using csipsimple silk integration did you applied the patch
>> that adds the silk rtp payload type?
>> Also do you get the same problem trying silk with a csipsimple build?
>>
>> BR
>> Regis
>>
>> Le 20 sept. 2012 06:57, "Alex" <alex.solis at telcentris.com> a ?crit :
>>>
>>> Hi,
>>>
>>> When using the SILK codec, I am only getting one way audio:
>>> rtp.c  pjmedia_rtp_session_init: ses=0x9bdc55c, default_pt=104,
>>> ssrc=0x55d47fd9
>>> rtp.c  pjmedia_rtp_session_init: ses=0x9bdc8a0, default_pt=121,
>>> ssrc=0x55d47fd9
>>>
>>> It looks like pjmedia is changing the payload type causing a mismatch at
>>> the server.
>>>
>>> Is there a way to determine what is causing the payload type to change
>>> from 104 to 121?
>>>
>>> Thanks in advance,
>>> Alex.
>>>
>>>
>>>
>>>
>>>
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>
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