[OpenSIPS-Users] SIP to WebRTC via OpenSIPS mid-registrar fails: forced proto 6 not matching sips uri

[Mark Allen]

Hi Juan Carlos - I feel your pain!

I've finished for the year so I don't have access to the code at the
moment. I'll have a look at this in the new year to see if I can post
something sensible! :-)

Short version is that it turned out to be a horrible challenge to get it
working as desired. My memory is that the difficulty in finding a suitable
workaround arose because of how OpenSIPS works (LUMPS system) when you make
changes in that they are only committed at the end. This means that if you
change $du to something new but if you then immediately xlog("$du") you'll
still see the original value - the change is only committed when you've
finished working on that SIP message.

In the WSS and mid-registrar case, it means that to workaround the wss
problem by using the path setting I had to loopback the SIP message to
OpenSIPS to then be able to see the Path value. I had to fudge some things
with code but it's working for us now. Maybe somebody else knows a more
straightforward way?????


On Wed, 23 Dec 2020 at 15:16, juancarlosg6 <juancarlosg6 at gmail.com> wrote:

> Hi, iam having the same situation, can you copy the opensips.cfg as
> reference, iam having issue with incoming calls to webrtc extensions, my
> opensips version is 3.1 and is working with mid_registar, i can do calls to
> SIP extensions, SIP Trunk, but not working when incoming calls to extension
> Webrtc is loged throung Opensips using mid_registar, thank you.
>
>
>
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