[asterisk-users] Some questions regarding jitterbuffer in asterisk / pjsip

[Michael Maier]

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