Hello, I'd appreciate your comments on the following problem I'm having, please forgive me if this is something obvious, I've been scratching my head on this for a while:
I have Asterisk+Kamailio setup where I'm currently testing inbound calls from outside. I have both webrtc and sip clients, where webrtc peers are defined according to sip.js instructions ( http://sipjs.com/guides/server-configuration/asterisk/). Calls between these work nicely without problems. Now when I call from outside, from an external Asterisk 11.5 server, I end up having problems calling from a sip client to a webrtc client. The Asterisk I have on my main testing server is the latest current 11.14.1. When there's an internal call, Asterisk changes the sdp in the INVITE message and handles the rtp nicely, but it does not do so when the call comes from outside. Why not? Instead, it sends back 488 Not acceptable here. If I react to that in Kamailio and use rtpengine to rewrite the sdp, Asterisk accepts the INVITE and sends it to the websocket peer, but the sdp contains a very simple sdp with RTP/AVP profile. This I'd consider invalid behavior, since Asterisk knows the called party is webrtc and the INVITE already contains valid sdp with RTP/SAVPF profile. It's likely I have something wrong in my setup, or maybe I've overlooked something relevant? Question is, what is causing this behavior and what can I do to fix it? Either I'd need Asterisk to handle the sdp and rtp like it does for internal calls (which would be preferable in this case) or after the 488 sent by Asterisk I'd need Asterisk to relay the sdp offered by Kamailio/rtpengine as such without rewriting it. Here the call works fine (internal call from sip peer 771 to webrtc peer 660): INVITE that Asterisk (at port 5070) receives: PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1046 INVITE sip:6...@testers.com;transport=UDP SIP/2.0 Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=41030177> Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bKd7b.ca8b6ac6a82d605cf658af0fea7c9e86.0 Via: SIP/2.0/UDP AST.ER.ISK.IP:38699;rport=38699;branch=z9hG4bK-d8754z-bd00e9fd46368417-1---d8754z- Max-Forwards: 69 Contact: <sip:7...@ast.er.isk.ip:38699;transport=UDP> To: <sip:6...@testers.com;transport=UDP> From: "771"<sip:7...@testers.com;transport=UDP>;tag=41030177 Call-ID: YWYwMjMwMmZlODEwM2MwODdjZWJmYjc2NjM5ZmIyNzk. CSeq: 2 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, X-cisco-serviceuri User-Agent: Z 3.2.21357 r21367 Allow-Events: presence, kpml Content-Length: 239 v=0 o=Z 0 0 IN IP4 AST.ER.ISK.IP s=Z c=IN IP4 AST.ER.ISK.IP t=0 0 m=audio 8000 RTP/AVP 3 110 8 0 98 101 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=20 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv Invite that Asterisk sends: PU.BL.IC.IP:5070 > PU.BL.IC.IP:5060: SIP, length: 1238 INVITE sip:6...@pu.bl.ic.ip:5060 SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK26a2386a;rport Max-Forwards: 70 From: "771 win8 minipc" <sip:7...@testers.com:5070>;tag=as05e60cc6 To: <sip:6...@pu.bl.ic.ip:5060> Contact: <sip:7...@pu.bl.ic.ip:5070> Call-ID: 7985f7161fcf1a6824b8388d451be...@testers.com CSeq: 102 INVITE User-Agent: I Am the Devil Date: Fri, 05 Dec 2014 15:50:33 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 663 v=0 o=root 777617621 777617621 IN IP4 PU.BL.IC.IP s=Asterisk PBX 11.14.1 c=IN IP4 PU.BL.IC.IP t=0 0 m=audio 15662 RTP/SAVPF 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=ice-ufrag:75a7e84431d15f682bd728ee10bd867d a=ice-pwd:028c19574216643c12188a8530f278f8 a=candidate:H5bdd423d 1 UDP 2130706431 PU.BL.IC.IP 15662 typ host a=candidate:H5bdd423d 2 UDP 2130706430 PU.BL.IC.IP 15663 typ host a=connection:new a=setup:actpass a=fingerprint:SHA-256 CE:EE:D9:28:EA:B0:6E:D0:CE:4F:5A:9A:FB:53:66:74:83:47:18:37:2F:76:C1:6D:10:C0:EE:FF:A4:56:F4:05 a=sendrecv Here the call fails (sip peer 201 calls from outside the server to webrtc peer 660): Invite that Asterisk receives: PU.BL.IC.IP:5060 > PU.BL.IC.IP:5070: SIP, length: 1345 INVITE sip:660%40testers....@pu.bl.ic.ip SIP/2.0 Record-Route: <sip:PU.BL.IC.IP;lr=on;ftag=as4647f03c;nat=yes> Via: SIP/2.0/UDP PU.BL.IC.IP;branch=z9hG4bK3264.8a896801756c527f2496fdc14e3f30ad.0 Via: SIP/2.0/UDP 192.168.0.201:5060 ;rport=5060;received=AST.ER.ISK.IP;branch=z9hG4bK56f5698e Max-Forwards: 69 From: "Pirjo Ahvenainen" <sip:201@192.168.0.201>;tag=as4647f03c To: <sip:660%40testers....@pu.bl.ic.ip> Contact: <sip:201@192.168.0.201:5060;alias=AST.ER.ISK.IP~5060~1> Call-ID: 69e66f05330de0063b5eba760191da6c@192.168.0.201:5060 CSeq: 102 INVITE User-Agent: Asterisk PBX 11.5.0 Date: Tue, 02 Dec 2014 08:34:57 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Type: application/sdp Content-Length: 547 v=0 o=root 1854132825 1854132825 IN IP4 PU.BL.IC.IP s=Asterisk PBX 11.5.0 c=IN IP4 PU.BL.IC.IP t=0 0 a=ice-lite m=audio 12516 RTP/AVP 8 3 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv a=rtcp:12517 a=ice-ufrag:5vj2cfWg a=ice-pwd:OdPg0e2qmExbbvAXUTLT3NI8g28s a=candidate:dEk0op4jY8ZkdPXr 1 UDP 2130706431 PU.BL.IC.IP 12516 typ host a=candidate:dEk0op4jY8ZkdPXr 2 UDP 2130706430 PU.BL.IC.IP 12517 typ host And the INVITE the Asterisk sends: PU.BL.IC.IP:5070 > PU.BL.IC.IP:5060: SIP, length: 847 INVITE sip:6...@testers.com SIP/2.0 Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK6f718231 Max-Forwards: 70 From: "771 win8 minipc" <sip:2...@pu.bl.ic.ip:5070>;tag=as2931af14 To: <sip:6...@testers.com> Contact: <sip:2...@pu.bl.ic.ip:5070> Call-ID: 04a3975e3bc84e6e32bfdc1905791...@pu.bl.ic.ip:5070 CSeq: 102 INVITE User-Agent: I Am the Devil Date: Fri, 05 Dec 2014 15:52:18 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 283 v=0 o=root 1272725383 1272725383 IN IP4 PU.BL.IC.IP s=Asterisk PBX 11.14.1 c=IN IP4 PU.BL.IC.IP t=0 0 m=audio 11906 RTP/AVP 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv And how are my peers configured? The calling peer on the separate Asterisk server is configured quickly in sip.conf: [general] bindport = 5060 bindaddr = 0.0.0.0 tcpbindaddr = 0.0.0.0 tcpenable = yes limitonpeers = yes rtcachefriends = no tos_sip = cs3 tos_audio = ef [201] type = friend secret = myverysecretpassword context = mytestcontext callerid = "201 User" <201> host = dynamic port = 5060 disallow = all allow = alaw allow = ulaw allow = gsm qualify = yes nat = no canreinvite = no and the dial command is like this: exten => 666,n,Dial(SIP/PU.BL.IC.IP/6...@testers.com) And the called number 660 is in the realtime sippeers table: id: 4 type: friend name: 660 host: dynamic secret: NULL encryption: yes avpf: yes icesupport: yes ipaddr: PU.BL.IC.IP port: 5060 regseconds: 1417782175 defaultuser: 660 fullcontact: sip:6...@pu.bl.ic.ip:5060 lastms: 0 useragent: context: mytestnumbercontext directmedia: no deny: 0.0.0.0/0.0.0.0 permit: PU.BL.IC.IP nat: force_rport,comedia transport: ws,wss,udp language: NULL disallow: NULL allow: NULL force_avp: yes callerid: 660 test amaflags: NULL mailbox: NULL regexten: NULL regserver: fromdomain: testers.com videosupport: no contactpermit: NULL contactdeny: NULL fullname: 660 win8 subscribemwi: NULL dtlsenable: yes dtlsverify: no dtlscertfile: /etc/asterisk/keys/asterisk.pem dtlsprivatekey: /etc/asterisk/keys/asterisk.pem dtlssetup: actpass sippasswd: a84a4ddcda13d13c9573d5294047b6a2 rpid: NULL domain: testers.com sippasswd2: 5c4671ae1043e6116118fed39bee091a callbackextension: NULL insecure: NULL cheers, Olli
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