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
--
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at: https://community.asterisk.org/
New to Asterisk? Start here:
https://wiki.asterisk.org/wiki/display/AST/Getting+Started
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users