VoIP Mailing List Archives

[krice at suspicious.org]

1. you need to include an SDP or enable 3pcc support on the sip profile...
2. the 3rd invite looks like it doesn’t include any codecs (or atleast any I am familiar with)
   

From: Adeel Ansari <adeel.gnome@gmail.com>
Reply-To: <freeswitch-users@lists.freeswitch.org>
Date: Tue, 11 Nov 2008 17:29:50 +0800
To: <freeswitch-users@lists.freeswitch.org>
Subject: [Freeswitch-users] Receiving 406 From Freeswitch....Any Clues?

Hi, I am getting 406, no matter what. I have tried 3 different INVITEs.

------
INVITE sip:1002@192.168.253.101 <mailto:sip%3A1002@192.168.253.101> ([email]sip%3A1002@192.168.253.101[/email]) ;transport=udp SIP/2.0
Call-ID: ef1e16976ae32f0f011de0db2ab5804b@192.168.253.101
CSeq: 1 INVITE
From: <sip:1001@192.168.253.101 <mailto:sip%3A1001@192.168.253.101> ([email]sip%3A1001@192.168.253.101[/email]) >;tag=5919
To: <sip:1002@192.168.253.101 <mailto:sip%3A1002@192.168.253.101> ([email]sip%3A1002@192.168.253.101[/email]) >
Via: SIP/2.0/UDP 192.168.253.101:7620;branch=z9hG4bKcfac6aaa7ee4a96457335bdad787cf31
Max-Forwards: 2
Contact: <sip:1001@192.168.253.101:7620;transport=udp>
Accept: audio/gsm,audio/x-gsm,text/plain
Content-Length: 0
------

and I tried,

-------
INVITE sip:1002@192.168.253.101 <mailto:sip%3A1002@192.168.253.101> ([email]sip%3A1002@192.168.253.101[/email]) ;transport=udp SIP/2.0
Call-ID: 7a612541e08e583e0069c988c8323425@192.168.253.101
CSeq: 1 INVITE
From: <sip:1001@192.168.253.101 <mailto:sip%3A1001@192.168.253.101> ([email]sip%3A1001@192.168.253.101[/email]) >;tag=7387
To: <sip:1002@192.168.253.101 <mailto:sip%3A1002@192.168.253.101> ([email]sip%3A1002@192.168.253.101[/email]) >
Via: SIP/2.0/UDP 192.168.253.101:6453;branch=z9hG4bK7ccbc7f8e7bc0c4ac0483cca54a48489
Max-Forwards: 2
Contact: <sip:1001@192.168.253.101:6453;transport=udp>
Accept: audio/gsm,audio/x-gsm,text/plain
Authorization: Digest username="1001",realm="192.168.253.101 <http://192.168.253.101> ",uri="sip:192.168.253.101:5060 <http://192.168.253.101:5060> ",algorithm=MD5,opaque="",nonce="b1f5e8d8-afd0-11dd-ad17-f7c4d6a988fd",response="2c70f260de2292d9e01406bdd1f90e28"
Content-Length: 0
-------

and then I tried,

------
INVITE sip:1002@192.168.253.101 <mailto:sip%3A1002@192.168.253.101> ([email]sip%3A1002@192.168.253.101[/email]) ;transport=udp SIP/2.0
Call-ID: 12e7c3af8d94ba048789f75e7036ea74@192.168.253.101
CSeq: 1 INVITE
From: <sip:1001@192.168.253.101 <mailto:sip%3A1001@192.168.253.101> ([email]sip%3A1001@192.168.253.101[/email]) >;tag=6309
To: <sip:1002@192.168.253.101 <mailto:sip%3A1002@192.168.253.101> ([email]sip%3A1002@192.168.253.101[/email]) >
Via: SIP/2.0/UDP 192.168.253.101:4487;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd
Max-Forwards: 2
Contact: <sip:1001@192.168.253.101:4487;transport=udp>
Accept: audio/gsm,audio/x-gsm,text/plain
Authorization: Digest username="1001",realm="192.168.253.101 <http://192.168.253.101> ",uri="sip:192.168.253.101:5060 <http://192.168.253.101:5060> ",algorithm=MD5,opaque="",nonce="85c12b22-afd2-11dd-ad17-f7c4d6a988fd",response="c8d6083112fc68b6e53d7e5b38e990eb"
Content-Type: application/sdp
Content-Length: 111

v=0
o=1001 279445 280814 IN IP4 192.168.253.101 <http://192.168.253.101>
s=-
c=IN IP4 192.168.253.101 <http://192.168.253.101>
t=0 0
m=audio 1436 RTP/AVP
------


But I always receive the same 406. Any idea. The response I am receiving from Freeswitch is,

-------
received response : SIP/2.0 406 Not Acceptable
Via: SIP/2.0/UDP 192.168.253.101:4487;branch=z9hG4bK3282799ef902293b7a4795d1cc3d78bd
From: <sip:1001@192.168.253.101 <mailto:sip%3A1001@192.168.253.101> ([email]sip%3A1001@192.168.253.101[/email]) >;tag=6309
To: <sip:1002@192.168.253.101 <mailto:sip%3A1002@192.168.253.101> ([email]sip%3A1002@192.168.253.101[/email]) >;tag=H1DH6S41c0U2N
Call-ID: 12e7c3af8d94ba048789f75e7036ea74@192.168.253.101
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9256M
Accept: application/sdp
Accept-Encoding:
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE,REGISTER,INFO,PUBLISH
Supported: 100rel,timer,precondition,path,replaces
Allow-Events: talk,presence,dialog,call-info,sla,include-session-description,presence.winfo,message-summary
Content-Length: 0
-------

Thanks.

--
Best,
Adeel Ansari

http://www.linkedin.com/in/adeelansari

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[adeel.gnome at gmail.com]

Actually, I am just trying to build a minimal client/UA for Freeswitch. Do I need to go into 3pcc or something. I remember I did play with 3pcc but at that time I was implementing a sip stack and trying to come up with a minimal softswitch. Am I misunderstanding something?

It sounds like, I need to include codecs. I have tried the similar INVITE with a Voip provider directly, and send it with 0 content-length, and no sdp at all. In response I got whole lot of things like codec, and all. Therefore, I did the same for the Freeswtich too. However, I will try including codecs.

Thanks.

On Tue, Nov 11, 2008 at 5:38 PM, Ken Rice <krice@suspicious.org (krice@suspicious.org)> wrote:

[adeel.gnome at gmail.com]

I managed to get rid of that 406. Thanks, Ken. I omitted ,Accept Header, this time.
Now stuck some elsewhere.

--------
send request:
INVITE sip:1002@192.168.253.101 ([email]sip%3A1002@192.168.253.101[/email]);transport=udp SIP/2.0
Call-ID: ed05ada0e9aeb009a7b24eccd0678577@192.168.253.101 (ed05ada0e9aeb009a7b24eccd0678577@192.168.253.101)
CSeq: 1 INVITE
From: <sip:1001@192.168.253.101 ([email]sip%3A1001@192.168.253.101[/email])>;tag=9782
To: <sip:1002@192.168.253.101 ([email]sip%3A1002@192.168.253.101[/email])>
Via: SIP/2.0/UDP 192.168.253.101:5073;branch=z9hG4bK1c1a8db63c4877132a77dacc5c931e99
Max-Forwards: 2
Contact: <sip:1001@192.168.253.101:5073;transport=udp>
Authorization: Digest username="1001",realm="192.168.253.101",uri="sip:192.168.253.101:5060",algorithm=MD5,opaque="",nonce="9b96327c-b094-11dd-b776-c14fb4e838dd",response="b425dde41cc093676705f0415344cf22"
Content-Type: application/sdp
Content-Length: 133

v=0
o=1001 920370 921739 IN IP4 192.168.253.101
s=-
c=IN IP4 192.168.253.101
t=0 0
m=audio 8756 RTP/AVP
a=rtpmap:0 PCMU/8000

received response:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.253.101:5073;branch=z9hG4bK1c1a8db63c4877132a77dacc5c931e99
From: <sip:1001@192.168.253.101 ([email]sip%3A1001@192.168.253.101[/email])>;tag=9782
To: <sip:1002@192.168.253.101 ([email]sip%3A1002@192.168.253.101[/email])>
Call-ID: ed05ada0e9aeb009a7b24eccd0678577@192.168.253.101 (ed05ada0e9aeb009a7b24eccd0678577@192.168.253.101)
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9256M
Content-Length: 0


received response:
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 192.168.253.101:5073;branch=z9hG4bK1c1a8db63c4877132a77dacc5c931e99
From: <sip:1001@192.168.253.101 ([email]sip%3A1001@192.168.253.101[/email])>;tag=9782
To: <sip:1002@192.168.253.101 ([email]sip%3A1002@192.168.253.101[/email])>;tag=gB110DNg8jt5Q
Call-ID: ed05ada0e9aeb009a7b24eccd0678577@192.168.253.101 (ed05ada0e9aeb009a7b24eccd0678577@192.168.253.101)
CSeq: 1 INVITE
User-Agent: FreeSWITCH-mod_sofia/1.0.trunk-9256M
Accept: application/sdp
Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,MESSAGE,SUBSCRIBE,NOTIFY,REFER,UPDATE,REGISTER,INFO,PUBLISH
Supported: 100rel,timer,precondition,path,replaces
Allow-Events: talk,presence,dialog,call-info,sla,include-session-description,presence.winfo,message-summary
Proxy-Authenticate: Digest realm="192.168.253.101",nonce="a40afd98-b094-11dd-b776-c14fb4e838dd",algorithm=MD5,qop="auth"
Content-Length: 0
--------

Although, I have included the authorization thingy in the INVITE request. Any idea, why its like that?

Thanks in advance.


On Tue, Nov 11, 2008 at 5:58 PM, Adeel Ansari <adeel.gnome@gmail.com (adeel.gnome@gmail.com)> wrote:

[mike at jerris.com]

To do a proper auth header we need to challenge you so you have the right nonce. After you have the nonce you can generate the proper auth header.

Mike

On Nov 12, 2008, at 4:27 AM, Adeel Ansari wrote: