1 May
2015
1 May
'15
10:44 p.m.
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 anybody confirm that he made successful browser to browser video
calls with Kamailio sip proxy / registrar in front of Asterisk PBX.
Also, any link to good tutorial or doc pages will be appreciated.
Best Regards,
Ivan Vujisic