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.org/message/WG44P2L4X5SLUG6HUJ3QQDI7R6G2RGPT/

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