15 Mar
2023
15 Mar
'23
7:19 p.m.
This is still the same thing that I already responded to on GH, as well
earlier on this mailing list:
https://lists.kamailio.org/mailman3/hyperkitty/list/sr-users@lists.kamailio…
You have no connectivity to your calling party. No ICE candidates are
provided in the offer, and no trickle ICE SDP fragments are passed to
rtpengine. The calling party could start making ICE/STUN requests to the
ICE candidate provided by rtpengine, but this never happens (no activity
on that port), possibly because it's a private address and may not be
reachable by the calling client.
On 13/03/2023 03.19, [EXT] Arun K R wrote:
Hello,
I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in
the same server. I have configured kamailio to work as webrtc server
and it forwards the registration to asterisk. Now when I am trying
call from jssip webrtc client it reaches kamailio and route it through
private interface to asterisk server. Asterisk then route it to the
provider server. When i make a call asterisk server recives rtp from
the provider and convert the rtp to srtp and sending back to kamailio.
but there is no sound for webrtc from public internet . Also i am
getting warning (SRTCP /RTP output wanted, but no crypto suite was
negotiated)
webrtc from Local network works fine with VPN
For normal udp call without webrtc works fine.
Kamailio having two interfaces, interface1 is private 10.13.1.140 and
interface 2 is publi ip 100.x.x.x
webrtc client sends calls to the public ip interface 100.x.x.x and
kamaiio routes the call to the asterisk via private interface 10.13.1.140.
asterisk server sends the call to remote server and gets the rtp back
from that
Asterisk server ---converts RTP to SRTP and forward to ----> kamailio
10.13.1.140
but there is nothing happens after that, please help me on this i am
new to kamailio and rtpengine.
The issue is only for webrtc from public internet. But when using
webrtc from LAN works fine
below are the logs