From rfuchs@sipwise.com Fri Mar 10 13:21:14 2023 From: Richard Fuchs To: sr-users@lists.kamailio.org Subject: [SR-Users] Re: webrtc from public internet have no voice Date: Fri, 10 Mar 2023 08:20:48 -0500 Message-ID: <5d0417f5-9cc6-6b9a-24f9-672518975f31@sipwise.com> In-Reply-To: <167837949559.1439.8136434817586636553@main.kamailio.org> MIME-Version: 1.0 Content-Type: multipart/mixed; boundary="===============1077879906==" --===============1077879906== Content-Type: text/plain; charset="utf-8" Content-Transfer-Encoding: quoted-printable As mentioned on GH, you have 1) rtpengine running on a private IP=20 address, so is not directly reachable from the public internet, and 2)=20 an offer from a WebRTC client using trickle ICE, which doesn't provide=20 any ICE candidates, and so that client isn't reachable from rtpengine.=20 Either give rtpengine a public address so that it can be reached from=20 outside, and/or (preferable "and") make sure the trickle ICE fragments=20 being sent by the WebRTC client (usually as SIP INFO) are passed to=20 rtpengine (and hoping that one of the candidates are a public address=20 that is reachable by rtpengine). Cheers On 09/03/2023 11.31, mail4arunkr(a)gmail.com wrote: > Hello, > I have installed Kamailio 5.6 in debian 11 and RTPengine 11.3 also in the s= ame server. I have configured kamailio to work as webrtc server and it forwar= ds 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 as= terisk server recives rtp from the provider and convert the rtp to srtp and s= ending back to kamailio. but there is no sound for webrtc from public interne= t . 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 inter= face 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 fr= om LAN works fine > > > > below are the logs > > Mar 9 20:21:02 debian /usr/local/sbin/kamailio[11778]: INFO: