Hi, just an idea. Put your call on hold, which means you can send audio out only while maintaining a standard call. BR, Andreas Am 19.08.2011 10:26, schrieb Michael Erskine: > Hi all, > > I'm successfully using the excellent PJSIP-UA library for PC-based > live audio public address: one-to-many outgoing only audio using > wideband speex (which gives amazing speech quality BTW!). Upon > establishing a call I enable only one-way audio inthe conference > bridge but I'm hoping to optimise my bandwidth usage by somehow > disabling the unneeded return RTP stream. Would this be easily done > and is it worth doing? I would like to maintain compatibility with > regular SIP phones and exchanges if possible! > > Regards, > Michael Erskine (msemtd). > > _______________________________________________ > 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 > > > ----- > eMail ist virenfrei. > Von AVG uberpruft - www.avg.de > Version: 10.0.1392 / Virendatenbank: 1520/3842 - Ausgabedatum: 18.08.2011 > >
