[SR-Users] Browser WebRTC transcoder

Thu May 19 10:52:15 CEST 2016

Yes G711 is offered... I guess the Grandstream phone is "confused" about the way the SIP browser stack talks. I got the following logs taken on Kamailio, using ngrep:

######################### START OF LOG ##########################
# Call from Grandstream to browser
##################################
[root at sip ~]# ngrep -d any -qt -W byline port 5060
interface: any
filter: ( port 5060 ) and (ip or ip6)
...
U 2016/05/19 09:15:24.701195 192.168.100.85:5060 -> 192.168.100.159:5060
INVITE sip:1001 at 192.168.100.159 SIP/2.0.
Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca.
From: "Fernando" <sip:1002 at 192.168.100.159>;tag=b0d53bed080e1b0f.
To: <sip:1001 at 192.168.100.159>.
Contact: <sip:1002 at 192.168.100.85:5060;transport=udp>.
Supported: replaces, timer, path.
P-Early-Media: Supported.
Proxy-Authorization: Digest username="1002", realm="192.168.100.159", algorithm=MD5, uri="sip:1001 at 192.168.100.159", nonce="Vz13SFc9dhwixIQaHqfrXSDvtNLkf+guJwNHi4A=", response="9c4f2fc2b7f179172fe9a6adf0d2f60f".
Call-ID: dda078a035a57ecb at 192.168.100.85.
CSeq: 8700 INVITE.
User-Agent: Grandstream BT200 1.2.5.3.
Max-Forwards: 70.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE,UPDATE,PRACK.
Content-Type: application/sdp.
Content-Length: 385.
.
v=0.
o=1002 8000 8001 IN IP4 192.168.100.85.
s=SIP Call.
c=IN IP4 192.168.100.85.
t=0 0.
m=audio 11022 RTP/AVP 18 8 0 3 9 2 97 101.
a=sendrecv.
a=rtpmap:18 G729/8000.
a=rtpmap:8 PCMA/8000.
a=rtpmap:0 PCMU/8000.
a=rtpmap:3 GSM/8000.
a=rtpmap:9 G722/8000.
a=rtpmap:2 G726-32/8000.
a=rtpmap:97 iLBC/8000.
a=fmtp:97 mode=20.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.

U 2016/05/19 09:15:24.710083 192.168.100.159:5060 -> 192.168.100.85:5060
SIP/2.0 404 Not Found.
Via: SIP/2.0/UDP 192.168.100.85:5060;branch=z9hG4bK402e7247e47576ca.
From: "Fernando" <sip:1002 at 192.168.100.159>;tag=b0d53bed080e1b0f.
To: <sip:1001 at 192.168.100.159>;tag=56f8047e80cfcc9e90a3acc9609da1ba-9e91.
Call-ID: dda078a035a57ecb at 192.168.100.85.
CSeq: 8700 INVITE.
Server: kamailio (4.2.3 (x86_64/linux)).
Content-Length: 0.

######################### END OF LOG ##########################

So the Grandstream offers a lot of codecs but will get a "Not Found" from Kamailio. Look in the other way:

######################### START OF LOG ##########################
# Call from browser to Grandstream
##################################
[root at sip ~]# ngrep -d any -qt -W byline port 5060
interface: any
filter: ( port 5060 ) and (ip or ip6)
U 2016/05/19 09:25:11.285826 192.168.100.159:5060 -> 192.168.100.85:5060
INVITE sip:1002 at 192.168.100.85:5060;transport=udp SIP/2.0.
Record-Route: <sip:192.168.100.159;r2=on;lr=on>.
Record-Route: <sip:192.168.100.159:4443;transport=ws;r2=on;lr=on>.
Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0.
Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318.
From: "Moacir"<sip:1001 at my.lab>;tag=MhRTswgaXENAxcDi25HJ.
To: <sip:1002 at my.lab>.
Contact: "Moacir"<sips:1001 at df7jal23ls0d.invalid;alias=192.168.100.249~59318~6;rtcweb-breaker=no;click2call=no;transport=wss>;+g.oma.sip-im;language="en,fr".
Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d.
CSeq: 15365 INVITE.
Content-Type: application/sdp.
Content-Length: 1182.
Max-Forwards: 69.
User-Agent: IM-client/OMA1.0 sipML5-v1.2016.03.04.
Organization: Doubango Telecom.
.
v=0.
o=mozilla...THIS_IS_SDPARTA-46.0.1 947803314240298800 0 IN IP4 127.0.0.1.
s=Doubango Telecom - firefox.
t=0 0.
a=sendrecv.
a=fingerprint:sha-256 44:F1:6D:31:F4:D6:D9:43:1D:38:0B:8E:67:1E:5F:DD:10:F4:5F:1C:4B:7E:7A:47:F8:85:C4:93:40:A7:2D:5E.
a=ice-options:trickle.
a=msid-semantic:WMS *.
m=audio 61455 UDP/TLS/RTP/SAVPF 109 9 0 8.
c=IN IP4 192.168.100.249.
a=candidate:0 1 UDP 2122252543 192.168.100.249 61455 typ host.
a=candidate:1 1 UDP 2122187007 2001:0:5ef5:79fd:24be:2fcf:fa06:dbc8 61456 typ host.
a=candidate:0 2 UDP 2122252542 192.168.100.249 61457 typ host.
a=candidate:1 2 UDP 2122187006 2001:0:5ef5:7
U 2016/05/19 09:25:11.348673 192.168.100.85:5060 -> 192.168.100.159:5060
SIP/2.0 488 Not Acceptable Here.
Via: SIP/2.0/UDP 192.168.100.159;branch=z9hG4bK5ffb.16431fc4d55b38465ed5bfedf4063ead.0.
Via: SIP/2.0/WSS df7jal23ls0d.invalid;received=192.168.100.249;branch=z9hG4bKTJM96ETVTs9QgSxvCWCBgWN1APMWKQVz;rport=59318.
Record-Route: <sip:192.168.100.159;r2=on;lr=on>.
Record-Route: <sip:192.168.100.159:4443;transport=ws;r2=on;lr=on>.
From: "Moacir"<sip:1001 at my.lab>;tag=MhRTswgaXENAxcDi25HJ.
To: <sip:1002 at my.lab>;tag=9ea71e1d1f839cef.
Call-ID: 2ecd7f42-ae98-563b-f0a7-00a4fe94c62d.
CSeq: 15365 INVITE.
User-Agent: Grandstream BT200 1.2.5.3.
Warning: 304 GS "Media type not available".
Content-Length: 0.

######################### END OF LOG ##########################

Here the Grandstream says "Media type not available". As I am not a real SIP guy, I got no clue why does not work!

Anyway, I am using the latest RPMs from Kamailo, running it using the websocket.cfg suggested configuration, no rtpengine installed on it. At the WebRTC, I am using sipml5 configuring it not to use STUN/TURN.

Cheers!
Moacir

To: sr-users at lists.sip-router.org
From: miconda at gmail.com
Date: Thu, 19 May 2016 06:22:57 +0200
Subject: Re: [SR-Users] Browser WebRTC transcoder

    What codecs are supported by your grandstream? Isn't the g711 in
      the group?

    Cheers,

      Daniel

    On 19/05/16 01:51, Moacir Ferreira
      wrote:

      I did not dig into the problem but on my tests I
        saw that my (old) Grandstream phone was refusing the call for
        not having a compatible codec to talk with the offered ones by
        the browser (Firefox). Being this the case, I guess I must
        include a translator, and all routing logic, in between the
        callers. It points to Asterisk that I would like to avoid for
        now. But I guess this is not a problem that only affects me.
        Someone else must have faced this before. So the question still
        open: What solution would be recommended for such case?

        Cheers,

        Moacir

        > To: sr-users at lists.sip-router.org

          > From: rfuchs at sipwise.com

          > Date: Wed, 18 May 2016 19:03:10 -0400

          > Subject: Re: [SR-Users] Browser WebRTC transcoder

          > 

          > On 18/05/16 04:57 PM, Moacir Ferreira wrote:

          > > Hey Daniel,

          > >

          > > If you say so, you probably right... I did not try
          it because on the

          > > sipwise GitHub
          (https://github.com/sipwise/rtpengine) they mention:

          > >

          > > /"Rtpengine does not (yet) support:/

          > > //

          > >

          > > * /Repacketization or transcoding/

          > 

          > This refers to translating one audio codec into another
          (e.g. opus to 

          > PCM). Translating between RTP and SRTP (i.e. encrypting
          and decrypting) 

          > is supported.

          > 

          > Cheers

          > 

          > _______________________________________________

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    -- 
Daniel-Constantin Mierla
http://www.asipto.com
http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
Kamailio World Conference, Berlin, May 18-20, 2016 - http://www.kamailioworld.com

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