Re: [SR-Users] Webrtc: Don't catch 488 between JSSIP and SIP UA's

2014-06-06 Thread LAA
Hi all, Another attempt, After doing some tests, I saw that one of the problems was that was necessary to comment the following lines within the deffinition of the RELAY route: # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. #if

[SR-Users] Webrtc: Don't catch 488 between JSSIP and SIP UA's

2014-06-02 Thread LAA
Apologize. Previous message was too long. L. El 02/06/2014 20:25, LAA ornitorrinco7...@gmail.com escribió: Hi all, Another guy strugling his mind trying to get a configuration to enable calls between WebRTC UA (JSSIP) to standard SIP UA (Twinkle or SjPhone) I've been working with the