16 May
2014
16 May
'14
9:45 a.m.
Hello!
During a call from classical SIP softphone to WebRTC there's no media from
the browser (Mozilla, the same result is for Chrome). In case of a call
from the browser to the softphone there's media flow from both sides.
The snippets from kamailio.cfg related to the problem case (SIP-->WebRTC)
are below.
OFFER:
$var(rtpp_flags) = "trust-address symmetric replace-origin
replace-session-connection";
$var(rtpp_flags) = $var(rtpp_flags) + " ICE=force";
$var(rtpp_flags) = $var(rtpp_flags) + " RTP/SAVPF";
rtpengine_offer($var(rtpp_flags));
ANSWER:
$var(rtpp_flags) = "trust-address symmetric replace-origin
replace-session-connection";
$var(rtpp_flags) = $var(rtpp_flags) + " ICE=remove";
$var(rtpp_flags) = $var(rtpp_flags) + " RTP/AVP";
rtp.log is attached.
Any help on this issue would be very appreciated.
with best regards,
Alexey