----- "Ondrej Valousek" <[email protected]> escreveu: > 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?
You're right. I just found a webpage that explains in detail the way the jitter buffer works: http://www.asterisk.org/node/48317. _______________________________________________ -- 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
