One way audio with SILK

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Hi Regis,

Thanks for you reply.

The implementation I have is based on csipsimple but for iPhone, so I cannot test a csipsimple build.

I do have the payload type in the types.h file but the enum is in a different order, in my case SILK 16KHz is currently 121 (this is the preferred codec I am using). What I don't understand is were pjsip is initially getting 104 as the payload type.

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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