[asterisk-users] Some questions regarding jitterbuffer in asterisk / pjsip
m1278468 at mailbox.org
Mon May 8 10:37:36 CDT 2017
I just implemented a jitterbuffer for pjsip in the dialplan in a SBC:
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!
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