Re: Help Adding WebRTC to Existing SIP Setup

Yuriy G via sr-users <[email protected]>
Newsgroups gmane.comp.voip.ser
Message-ID <CABSP_VdsWitaG5usv=GBC9q3aRmUbOum=QB+ySX2hnzSntWqwg@mail.gmail.com>
Yes that is possible.
You can use this one as reference

https://github.com/havfo/WEBRTC-to-SIP

On Thu, May 22, 2025, 19:16 Fernando Lopes via sr-users <
[email protected]> wrote:

> Hi everyone,
>
> I currently have a working SIP setup, but I'm trying to get WebRTC working
> alongside it and haven’t had much success so far. Does anyone have tips or
> best practices for enabling both SIP and WebRTC in the same configuration?
> Is it possible to handle both within a single Kamailio config? Or would I
> need some kind of conditional logic (like if/else) to manage how the config
> handles different types of clients?
>
> My setup looks like this:
> Router → Kamailio → Kubernetes (Asterisk inside the cluster)
> I was having some issues with the Contact header, but I managed to resolve
> that. Now I'm stuck figuring out how to make WebRTC work cleanly alongside
> SIP.
> Any advice would be appreciated!
>
> Thanks!
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