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 :-)
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:
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,
you also be aware of longer callsetups to webrtc clients (13 - 38 seconds) if there are many ICE candidates on the client (like different networking devices).
There maybe also fixes in the js SIP libraries for that (I don't touch this area).
Am 05.01.2018 9:04 vorm. schrieb "Daniel-Constantin Mierla" < miconda@gmail.com>:
Hello,
On 05.01.18 09:55, Karsten Horsmann wrote:
if you know that it is low chance to work without a rtp relay (such as rtpengine), one option here is to remove all ice candidates and let only the ones from the relay. This is a solution in kamailio.cfg.
Otherwise, there are some variants of ICE that should make the setup faster (e.g., use relay first, they try other better variants and if one works switch to it). Not sure the state of implementation in browsers and the options offered by various sip js libraries.
Cheers, Daniel
-
Daniel-Constantin Mierla -
Karsten Horsmann -
paul＠kristianpaul.org