If pjsip path doest work ( which indeed can be a case ) It is an option for you to mascarade contact on kamailio ( if you need to register phones on asterisk ), but this is not a trivial.
If you do registrar on kamailio then lookup and set up proper $du for webrtc endpoints will workout for you I believe. On Thu, 6 May 2021, 20:43 Eliphas Levy Theodoro, <elip...@gmail.com> wrote: > 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 >
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