16 May
2014
16 May
'14
8:09 a.m.
Hello Alexey,
I am uncertain as to whether rtpengine_offer()/answer() support
pseudovariable arguments. But if they do, you'll need to wrap them in a
string literal:
rtpengine_offer("$var(rtpp_flags)");
If they don't support PV arguments at all, you may be stuck with having
to provide a static argument:
rtpengine_offer("trust-address symmetric replace-origin
replace-session-connection ICE=force RTP/SAVPF");
-- Alex
On 05/16/2014 02:45 AM, Alexey Rybalko wrote:
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
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--
Alex Balashov - Principal
Evariste Systems LLC
235 E Ponce de Leon Ave
Suite 106
Decatur, GA 30030
United States
Tel: +1-678-954-0670
Web: http://www.evaristesys.com/, http://www.alexbalashov.com/