[asterisk-users] chan_sip and matching the RTP source

[Dovid Bender]

All,

I am stuck with a specific install using chan_sip and Asterisk 11.25.3. We
have nat=no which from what I understand means that Asterisk will go by
whatever it see's in the SDP and not look at the source IP+port from where
the traffic is coming from. We have a call flow where we send a carrier a
call and they specify an IP and port in their SDP in a 183 (e.g.
100.100.100.100:36070). As we get that 183 the carrier starts sending RTP
from that IP and port so Asterisk does the same and sends rtp back to that
IP and port. Two seconds later we get RTP from a new IP and port (e.g.
200.200.200.200:21592). Two seconds after that we get an updated 183 with
SDP and then new IP and port. Asterisk keeps sending media to the old IP
(100.100.100.100). We then get a 200 OK and then again in the SDP there is
the new IP and port. Asterisk keeps sending rtp to the old port and IP.
>From what I understand if we have nat=no then we should update where we
send RTP and send it to the IP in the SDP and not look at the source Is
that correct or am I wrong? Is there any other setting that I can be
missing?

TIA.

Dovid
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20201023/1c0463c5/attachment.html>