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).
