Hello,
kamailio + rtpengine can be used for webrtc calls between browsers as
well as browser to classic sip phones. You can fine on github some
config examples, published by Carlos Ruiz Diaz.
Using this combination you can place an instance in front of asterisk
and let asterisk behave as a classi
I made successful audio calls from browser to browser using Asterisk
13.1 and SIPML5 browser phone.
Asterisk can't manage WebRTC video calls due to lack of codec
negotiation module, but I also faced RTP ports NAT traversal issue. To
my understanding Kamailio is capable to resolve this.
Can anybo
