[OpenSIPS-Users] Change in SDP/RTP routing with Opensips 1.4 and 1.6

[Ovidiu Sas]

You din't posted a proper SIP capture (no UDP src/dst IP for your
packets), but it seems that you are having a firewall issue.  Just add
an xlog into your opensips server and see that you will have no
entries in syslog.
The INVITE that is sent out is having the same "Max-Forwards" counter
and you have a second Via with exactly the same branch parameter.


Regards,
Ovidiu Sas

On Tue, Feb 8, 2011 at 3:58 PM, Chris Stone <axisml at gmail.com> wrote:
> Ovidiu,
>
> On Tue, Feb 8, 2011 at 1:16 PM, Ovidiu Sas <osas at voipembedded.com> wrote:
>> Based on your description, it seems that you are dealing with a weird
>> firewall/NAT that is SIP aware.
>> Also, I don't know if this is a typo, but you are forwarding with
>> opensips to 67.212.153.179 and your INVITE is actually sent to
>> 67.212.153.178.
>>
>> Try to turn off the firewall and NAT ans see it this is fixing your issue.
>
> No firewall - though of that too at first. There is a PIX between the
> net and the servers, but not between the Opensips server and the
> Asterisk server. The packet capture showing the modifications to the
> INVITES was from the Opensips server - in-n-out. No firewall involved
> at that point.
>
> The Opensips server is 67.212.153.178 and it's forwarding to the
> Asterisk server which is at 67.212.153.179.
>
> Due to previous issues with the sip fixup in PIX firewalls, that it
> disabled in the PIX we're using. But again, this modification is being
> done after the PIX hands it off to the Opensips server....
>
>
> Thanks!
>
>
> Chris
>
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