23 Sep
2005
23 Sep
'05
6:17 a.m.
Ray,
Jitter buffers in VoIP are only of use at the point where the packets are
converted into the final PCM bit stream - a TDM or DAC interface in most
cases.
The problem with your proposed solution is that the output of mediaproxy,
even if it is de-jittered, will be subject to jitter once again on the way
to the endpoint where the packet connection is converted to voice.
Mark
On 9/21/05, Ray Van Dolson <rayvd(a)digitalpath.net> wrote:
First off, here is my voice network layout currently:
http://webdev.digitalpath.net/~rayvd/voice/voice_network2.png
We're using Asterisk for voicemail, call routing (for long-distance, LNP,
etc)
and SER/rtpproxy at the other end which handles NAT onto private networks
where customer's exist.
This setup works fairly well for the most part, except that Asterisk does
not
have a jitter buffer. I would like to make use of rtpproxy (or mediaproxy)
for their jitter buffer on both ends of our voice links here. To me, that
would mean shoving a SER/rtpproxy combo between Asterisk and our provider
network. Possibly on the same server.
I could easily throw up a SER installation, but I'm trying to figure out
if
there's any way to leave Asterisk in the SIP path but remove it from the
media
path (have RTP just go straight from SER/rtpproxy to my provider's RTP
proxy).
Have any of you set up a scenario somewhat like this? Any recommendations?
Asterisk CVS-HEAD does let you apply a patch and get some very alpha
jitter
buffer support. But CVS HEAD doesn't work reliably for me at all currently
so
I'm sticking with the latest stable release of 1.0.x series.
I'm running SER 0.9.3 FYI on the SER proxies I have set up currently.
Thanks.
--
Ray Van Dolson
Linux/Unix Systems Administrator
Digital Path, Inc.
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