Comments (4)
This project is not meant as an advanced routing configuration, but more of an example for translating between WebRTC and SIP. There are plenty of examples for using dispatcher available, I even have another repository that uses dispatcher here.
from webrtc-to-sip.
I tried to configure it, but I'm not an expert, can we go to email, irc, jabber? Would you like to help for a small fee?
from webrtc-to-sip.
@furek1 the learning curve on Kamailio is high. You should read the module documentations and try to adapt the configs in your own lab. If you then have problems you can make qualified questions on the Kamailio mailing list.
Also on the Kamailio website there is an business directory of consultants if you are looking for paid services for configuration and planing.
This Github repo is an excellent guide to adapt webrtc to SIP with Kamailio and stuff around it.
Just my thoughts about that.
from webrtc-to-sip.
@khorsmann The mailing list is poor, because no one talks there anymore about the dedicated channel :(
from webrtc-to-sip.
Related Issues (20)
- Build doesn't work HOT 2
- SIP Client with WebRTC SDP HOT 1
- Parallel call forking case HOT 6
- Error!! Unmet build dependencies : libbcg729-dev
- E: Unable to correct problems, you have held broken packages. HOT 2
- When lets encrypt permission denied
- No audio from SIP to WS HOT 2
- Integration with Asterisk HOT 1
- NO_PUBKEY Error in debian 10 HOT 1
- Testtest
- could not login sipml5 demo to kamailio via websocket. HOT 3
- Sip WIse installation in ubuntu 18.04 HOT 1
- Kamailio + RTPEngine + TURN server behind NAT HOT 5
- webrtc solution for sip protocol
- SIP to WebRTC calls only one way audio HOT 5
- Dockerfile ?
- why use turn server, can I only use rtpengine? HOT 3
- E: Package 'ngcp-keyring' has no installation candidate
- Audio issues from Webrtc to SIP HOT 3
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