Hi,
I am aware of the difference between media and signalling as well as the way an rtp proxy works. But still do I think that this is a different case:
When using asterisk, xlite does symetric rtp streams aka. sends from port 8000 and listens on port 8000. The port is therefore open on nat and udp pakets can travel through the nat with the correct public ip in the sip messages. When I do have the setup with ser and the sip messages look the same, aka. xlite says "I am listening on 8000 on public ip" and send via 8000 to the cisco phone which says "I am listening on port x on my ip". So there should be no need for an rtp proxy as the rtp stream from xlite is symetric and the nat port is open. But still there is no incoming audio. I can understand that with clients like MS messenger there is a need to rewrite the sdp and sip messages.
So where is the problem in my szenario, what little magic does asterisk do that I do not see in the sip messages???
Cheers Morten
SER is simply a proxy - it does not handle media in the same way that asterisk does. When you have both clients registered with SER, once the initial call set up has been completed, no further traffic runs through SER. Search this list for explanations as to why RTP traffic doesn't really run through NAT without a helping hand.
If you want to be able to make calls without any special client/NAT router settings, check out RTPproxy/NAThelper and Mediaproxy - they do the RTP proxy bit that Asterisk has built in.
Hope this helps.
Dave