[asterisk-users] Differences between Chan_SIP and PJSIP with NAT and STUN
Joshua Colp
jcolp at digium.com
Mon Mar 7 15:41:56 CST 2016
Chirag Desai wrote:
>
>
>
> Joshua Colp wrote:
>
> Have you done a packet capture to see if the RTP from the remote device
> is hitting the machine to narrow things down?
>
>
>
> Nope. When I run with RTP encryption on it seems that rewrite_contact
> does not work in PJSIP.
>
> When I turn off RTP some calls get media, some don't. If you look at the
> SIP trace it seems like the rewrite_contact doesn't always take affect.
The rewrite_contact shows as working fine in the SIP trace. The log
shows the message as received over the socket, before modification. If
it wasn't working then the BYE would be going to the internal IP
address+port.
Nothing stands out in the signaling.
--
Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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