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? 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. 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. Many thanks for all answers, I have tried hard to google out them, but no success so far. Ondrej
----- "Ondrej Valousek" <webserv at s3group.cz> 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-usersVin?cius Fontes www.asteriskforum.com.br - Informa??es e discuss?o sobre Asterisk e telefonia IP
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