Hello, I now understood that to send the request directly to the socket instead of via UDP/invalid.ip.address. Succeded by lookup() after save(), or asterisk's, Dial(PJSIP/local_aor) if registered locally.
I settled with passing registration by domain to the right asterisk server beneath kamailio, and save()ing on reply 200 OK. On the asterisks, I set kamailio as outbound proxy; They then dial the sip:aor@domain and kamailio will lookup($rd) easily. By the way, why save("$rd") and lookup("$rd") won't substitute for request uri domain? It is saving a literal $rd on the location table. In this case I will need to have multiple if()s with each domain name. Should I use another function/table for saving multidomain contacts? Now that the signaling is doing alright, I need to figure out about the media, starting with that config from gitub/havfo (WEBRTC-to-SIP). Thanks, Eliphas Em qui., 6 de mai. de 2021 às 16:03, Yuriy Gorlichenko <ovoshl...@gmail.com> escreveu: > > 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 > > __________________________________________________________ > 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 __________________________________________________________ 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! 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