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[ohjelmistoarkkitehti a...]

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:660@testers.com ([email]sip%3A660@testers.com[/email]);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:771@AST.ER.ISK.IP:38699;transport=UDP>
To: <sip:660@testers.com ([email]sip%3A660@testers.com[/email]);transport=UDP>
From: "771"<sip:771@testers.com ([email]sip%3A771@testers.com[/email]);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:660@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:771@testers.com:5070>;tag=as05e60cc6
To: <sip:660@PU.BL.IC.IP:5060>
Contact: <sip:771@PU.BL.IC.IP:5070>
Call-ID: 7985f7161fcf1a6824b8388d451bec16@testers.com (7985f7161fcf1a6824b8388d451bec16@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.com@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 ([email]sip%3A201@192.168.0.201[/email])>;tag=as4647f03c
To: <sip:660%40testers.com@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:660@testers.com ([email]sip%3A660@testers.com[/email]) SIP/2.0
Via: SIP/2.0/UDP PU.BL.IC.IP:5070;branch=z9hG4bK6f718231
Max-Forwards: 70
From: "771 win8 minipc" <sip:201@PU.BL.IC.IP:5070>;tag=as2931af14
To: <sip:660@testers.com ([email]sip%3A660@testers.com[/email])>
Contact: <sip:201@PU.BL.IC.IP:5070>
Call-ID: 04a3975e3bc84e6e32bfdc1905791913@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/660@testers.com (660@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:660@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

[mailinglist+asterisk a...]

On 05/12/14 16:46, Olli Heiskanen wrote:

[ohjelmistoarkkitehti a...]

Hello,

Thanks Gareth for your reply. I assume you're referring to the first INVITE in my message, which is from the call that works. I don't know why the sdp displays only iLBC and speex at that point but the Zoiper client that's making the call is configured to support gsm, speex, ulaw, alaw, and iLBC, and the call works fine, audio and all, as the sdp that leaves Asterisk (thus reaches the called peer) actually contains ulaw, gsm and alaw.


In the failing case Asterisk sends the INVITE via Kamailio to the called webrtc client, and in this message the rtp profile is m=audio 12902 RTP/AVP 0 3 8 101. Kamailio sends the INVITE to the client, which responds with 488. Kamailio notices this and uses rtpengine to handle the rtp, but: the client will not accept a second INVITE even though the sdp is correct this time: the client responds with 482 Loop Detected because the Call-ID is the same as the previous INVITE it got. This is why I can't handle the rtp using rtpengine, and here things have already gone wrong. So I need the INVITE to contain correct sdp when it leaves Asterisk, so sdp conversion and rtpengine would net be needed. Wonder if there's any way to do that?


cheers,
Olli








2014-12-05 18:53 GMT+02:00 Gareth Blades <mailinglist+asterisk@dns99.co.uk ([email]mailinglist+asterisk@dns99.co.uk[/email])>: