Hello,
another thing that one should be aware of is that in webrtc/websocket some sip headers (e.g., via, contact) use a random string instead of ip addresses and many old devices will throw parsing error. jssip (and maybe other js sip stacks) has an option to enable using a private ip address instead of a random ip address.
Cheers, Daniel
On 27.12.17 12:55, Karsten Horsmann wrote:
Hello Paul and List,
you can use the nice WebRTC Example from havfo at github https://github.com/havfo/WEBRTC-to-SIP
The magic with kamailio, rtpengine and WebRTC / SIP Bridging starts in this route.
https://github.com/havfo/WEBRTC-to-SIP/blob/master/etc/kamailio/kamailio.cfg...
You can merge this with the default kamailio advanced config to create an kamailio/rtpengine SBC.
Kind Regards
2017-12-26 14:35 GMT+01:00 <paul@kristianpaul.org mailto:paul@kristianpaul.org>:
Hi, I'm looking to use kamailio as a webrtc proxy for legacy sip system that doesnt have this capability, is there a example or blueprint i can follow to get started with this? I'm RTFMing the docs but still need a while to understand kamailio internals :-) _______________________________________________ Kamailio (SER) - Users Mailing List sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users <https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users>-- Mit freundlichen Grüßen *Karsten Horsmann*
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