Michael Maier
2017-May-08 15:37 UTC
[asterisk-users] Some questions regarding jitterbuffer in asterisk / pjsip
Hello! I just implemented a jitterbuffer for pjsip in the dialplan in a SBC: [fromtrunk] exten => _[+0-9]!,1,Set(JITTERBUFFER(fixed)=default) This jitterbuffer catches all calls coming from ISP. My understanding is, that the incoming rtp stream in leg1a is now buffered and handed out "jitter-optimized" to leg2a on the other site (this could be internal or external again). -----------> leg1a leg2a ------------> ISP SBC callee <----------- leg1b leg2b <------------ My question: What's about the rtp stream which is received by leg1b from callee? Is there a receive buffer on the leg1b-site, too? Or is it expected to be done by leg2b before handing it out to leg1b? Iow: is it enough to implement one jitterbuffer? Or should there be a second jitterbuffer on the side of leg2? Thanks for clarification! Regards, Michael