Ray, Just looking at your diagram again it seems puzzling as to why the lack of a jitter buffer in Asterisk would be a problem for you at all. You are not using Asterisk as a gateway. Mark
On 9/21/05, Ray Van Dolson rayvd@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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