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! > __________________________________________________________ > Kamailio - Users Mailing List - Non Commercial Discussions -- > [email protected] > To unsubscribe send an email to [email protected] > Important: keep the mailing list in the recipients, do not reply only to > the sender! > __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions -- [email protected] To unsubscribe send an email to [email protected] Important: keep the mailing list in the recipients, do not reply only to the sender!