As I have got 4 different answers (thanks!) I will paste them all down there.
Em qua., 5 de mai. de 2021 às 18:44, Eliphas Levy Theodoro <elip...@gmail.com> escreveu: > > I am trying to config one kamailio as reverse proxy for a bunch of internal > (no internet address) separate asterisk sip > instances (per domain). The kamailio server would be the only with the valid > IP address, so would use rtpengine to > force to be in the media path. > > Like this scenario: > https://opensips.org/pipermail/users/2020-August/043610.html > > I have used as starting point this very basic config: > https://blog.voipxswitch.com/2015/03/27/kamailio-basic-sip-proxy-all-requests-setup/ > > Basically just added rtpproxy support, and voilà, inter-SIP is working, media > always passing into the proxy. > > The problem: I would have WebRTC phones connecting too. I tried setting WSS > up in kamailio, and asterisk (pjsip) > wouldn't know how to send the message to the proxy: on register it has > trasnport=wss in the contact: header, looks > like it is confusing the asterisk. > > So, I resort for the wisdom of the list :) What would be the good-best-path > to take here, hack the header, or put the > webphones registering directly on the asterisks (with a nginx reverse proxy > maybe)? [..] Daniel-Constantin Mierla mico...@gmail.com por lists.kamailio.org 06:26 (há 8 horas) > > if both endpoints can do webrtc srtp, then it works with rtpproxy to do srtp > packet forwarding for nat traversal or networks bridging. Yes, when a pair of softphones (ok) and softphones (not yet) exchange signalling alright in that scenario, I will start on transcoding... Wojtko, Daniel daniel.woj...@rittec.de por lists.kamailio.org 05:32 (há 8 horas) > afaik rtpproxy doesn't support WebRTC but rtpengine does As Daniel said above, I reckon that rtpproxy would work when transcoding/translating sip/webrtc is not needed. But first, need to pass signalling at least :) Yuriy Gorlichenko ovoshl...@gmail.com por lists.kamailio.org 05:55 (há 8 horas) > > If you looking for examples: you can use this one > https://github.com/havfo/WEBRTC-to-SIP as starting point > > anyway, the Path mentioned by Alex is the best approach. I tried that one but could not figure most of it out... I think I borked it. Tried only changing $du to asterisk instead of doing register locally and got the same results (and lots of rtpengine chattiness) too, so I am using now a very simple config to make finding the signalling problem easier. > Alex Balashov abalas...@evaristesys.com por lists.kamailio.org 03:26 (há 10 > horas) > It sounds like you are in need of the Path extension: That was one of the modifications I have made, found out later that the problem is PJSIP not handling Path: anyway: https://community.asterisk.org/t/pjsip-path-module-issues/88046 https://issues.asterisk.org/jira/browse/ASTERISK-28211 So I have changed back to the older chan_sip interface, problem solved, that one worked with Path: header. Now asterisk sends the invite back to kamailio! Now, the basic signalling of webphone -> kamailio -> asterisk -> kamailio -> otherphone is stopping on kamailio itself, it is sending the packet via UDP like asterisk was, instead of using the socket. This is how the webphone contact looks like: <sip:cakrtk0i@192.0.2.210;transport=wss> Kamailio (and asterisk before Path: worked) invites UDP:192.0.2.210:5060, instead of the "local" websocket, and of course never succeeding. I tried save()ing the register locally, but I am sure I am doing it wrong. if someone wants to look at the actual test config, I pasted it: https://pastebin.com/RuXniDTU Cheers, -- Eliphas __________________________________________________________ Kamailio - Users Mailing List - Non Commercial Discussions * sr-users@lists.kamailio.org Important: keep the mailing list in the recipients, do not reply only to the sender! Edit mailing list options or unsubscribe: * https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
