Hi Vinicius. >>/ 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. Ok, thanks!
>>/ 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. Yes, that makes a sense. I was just wondering if it is possible to configure it the way that the jitterbuffer is enabled only if the asterisk server can not do native RTP bridging... >>/ 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]. Well, I think that would not work since the jitterbuffer is only effective on the outgoing channels. If I receive a call from the SIP trunk, I hear jitter. To suppress it, I would have to enable jbforce/jbenable on my local SIP channel as this is the outgoing one - the SIP trunk is the incoming one, right? Many thanks, 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
