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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.pjsip.org/pipermail/pjsip_lists.pjsip.org/attachments/20120920/b2a9eaf6/attachment-0001.html>
