----- "Ondrej Valousek" <[email protected]> escreveu: > Hi List, > > I have a question regarding jitterbuffer in Asterisk 1.4.24. I see > that > jitterbuffer is only effective on the receiving channels. > My asterisk has only SIP accounts + 2 SIP trunk accounts to our branch > > office. > Questions: > 1. To enable jitter buffer on SIP channels it seems I have to enable > and > force it, right?
Not sure about the forcing part (don't know exacly how it works), but I always set jbforce=yes to be sure. > 2. If I enable and force jitter buffer, Asterisk would always have to > > stay in media path to make it function, right? If I am right, this > effectively disables native RTP bridging. Yes, there's no way Asterisk can create buffers if it's not on the media path. > 3. Is it possible to only enable jitter buffer on calls where the SIP > > trunk is involved? It is no use for me to enable the jitter buffer > between SIP phones on the same LAN. Sure, just put the jbenable and other options on the SIP section of that trunk, instead of putting it on [general]. > > Many thanks for all answers, I have tried hard to google out them, but > > no success so far. > Ondrej > > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users Vinícius Fontes www.asteriskforum.com.br - Informações e discussão sobre Asterisk e telefonia IP _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
