Leif Madsen wrote:
On Nov 19, 2007 8:22 PM, Ovidiu Sas <[EMAIL PROTECTED]> wrote:
I don't have it setup like that -- but if OpenSER doesn't have to talk
to something behind NAT, then yes -- either that, or from what I've
heard, you need to have a local proxy that everything goes through so
it can route it inside your internal network. I don't believe it can
do it if your company is behind NAT and all the phones are exposed as
only a single IP address.
Multiple phones behind the same NAT can be handled properly using the
openser/rtpproxy tandem.
Also, openser can easily handle Phone-to-Phone traffic, providing
features like call forwarding, CLID, speed dial, smart routing (lcr,
enum ...) and it can integrate with a cluster of asterisk boxes for
features like: voicemail, conferencing, transcoding (using the setup
that Leif described).

Does that actually keep the RTP stream behind their NAT, or does
OpenSER actually handle the RTP stream through the rtpproxy module?
Basically you're saying OpenSER handles the RTP stream instead of
Asterisk, correct?


You have both options here:

1. Detect that both clients are behind the same NAT and let the RTP streams flow directly between clients. An openser configuration example can be found here:
http://www.openser.org/dokuwiki/doku.php/examples:caller-callee-behind-same-nat
This may present issues if several private networks with or without NAT are behind the primary NAT.

 2. Handle the RTP stream through the rtpproxy server.
The disadvantage here is the bandwidth utilization.


In both cases, asterisk is out of the picture (if you do phone-openser-phone scenario). If asterisk is involved (phone-openser-asterisk-openser-phone) than it is a little bit more complex and I don't know for sure if scenario 1 is valid (asterisk may touch the SDP even for canreinvite=yes, but again, I'm not 100% sure).


Regards,
Ovidiu Sas

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