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. >> >> >> >> >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org >> >> _______________________________________________ >> Visit our blog: http://blog.pjsip.org >> >> pjsip mailing list >> pjsip at lists.pjsip.org >> http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org > > > _______________________________________________ > Visit our blog: http://blog.pjsip.org > > pjsip mailing list > pjsip at lists.pjsip.org > http://lists.pjsip.org/mailman/listinfo/pjsip_lists.pjsip.org -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120921/8a9670c6/attachment-0001.html>
