4 May
2015
4 May
'15
9:21 a.m.
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 classic sip/rtp media server.
Cheers,
Daniel
On 02/05/15 00:44, Ivan Vujisic wrote:
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
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Daniel-Constantin Mierla
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