That has been the case that the Invite is sent out without SDP.  It is my 
understanding that the receiver must accept that as a valid invite.   It seems 
that is a way to query the other side to see what they have to offer.

 

One ITSP that I have worked with had issues with that, but after showing them 
the argument for why they must accept that, they agreed they had an issue.  I 
believe the note below is relevant to your discussion.

 

Section 14.1 UAC Behavior:
The same offer-answer model that applies to session descriptions in INVITEs 
(Section 13.2.1) applies to re-INVITEs.
...a UAC MAY send a re-INVITE with no session description, in which case the 
first reliable non-failure response to the re-INVITE will contain the offer
(in this specification, that is a 2xx response).

 

 

From: [email protected] 
[mailto:[email protected]] On Behalf Of Sven Evensen
Sent: Thursday, June 07, 2012 6:05 AM
To: Joegen Baclor
Cc: Discussion list for users of sipXecs software
Subject: Re: [sipx-users] Transfer from attendant failing

 

Yes that I what I thought too. I will ask why they do that. I saw another SIP 
trunk sent "Restricted" and that "survived" and was passed on later in the URI 
and Contact fields.

 

But as you mentioned (double whammy) is that we send INVITE (for REFER) without 
SDP (why is that?) and they respond with a pretty much empty 200OK. they told 
me they dont support the late answer/offer.

 

So unless we can send SDP in that INVITE, they just dont support it and we are 
doomed.

 

Thanks for helping out Joegen!

 

Sven

On Thu, Jun 7, 2012 at 1:53 PM, Joegen Baclor <[email protected]> wrote:

sipXbridge did not invent that.  That is based on the contact they sent in the 
INVITE that created the dialog.  Pasting....

Contact: "Anonymous"<sip:194.6.238.85:5061>



On 06/07/2012 08:45 PM, Sven Evensen wrote: 

Sorry for being so vague. 

 

The URI of the INVITE is

> INVITE sip:194.6.238.85:5061 <http://194.6.238.85:5061/>  SIP/2.0 

 

On another system I have, identical scenarion, it looked like this (other IP 
address of course)

> INVITE sip:[email protected]:5061 <http://194.6.238.85:5061/>  SIP/2.0 

 

The 200OK coming back had then mod_sofia as in the contact field. I really dont 
know this in detail, so this is just observations from me.

 

Sven

On Thu, Jun 7, 2012 at 1:15 PM, Joegen Baclor <[email protected]> wrote:

Not sure what you mean by "because of our INVITE".  The INVITE we sent is 
correctly formatted.  pasting it again.  What do find wrong in the INVITE?

> INVITE sip:194.6.238.85:5061 SIP/2.0
> CSeq: 1 INVITE
> To: <sip:[email protected] 
> <mailto:sip%[email protected]> >;tag=rpz2agt27fobzvvd.o
> From: <sip:[email protected] <mailto:sip%[email protected]> 
> >;tag=55886164
> Call-ID: [email protected]
> Max-Forwards: 70
> Route: <sip:194.6.238.85;lr>
> User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:[email protected] 
> <mailto:sip%[email protected]> >
> References: [email protected].
> 101-0;rel=chain;sipxecs-tag=request-refer-z9hg4bk-xx-4ab4jjiscj
> `j9ig27d8fw74puw
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Supported: timer
> Via: SIP/2.0/UDP 176.34.136.92:
> 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836
> Contact: <sip:176.34.136.92:5080>
> Accept: application/sdp
> Content-Length: 0



On 06/07/2012 07:10 PM, Sven Evensen wrote: 

But I believe the reason they are sending no Contact field is because of our 
INVITE 

 

10600 21:21:22.433629 10.1.1.15 5080 194.6.238.85 5060 SIP Request: INVITE 
sip:194.6.238.85:5061, in-dialog

 

Do you know why the URI has no proper address?

On Thu, Jun 7, 2012 at 11:51 AM, Joegen Baclor <[email protected]> wrote:

Indeed.  200 OK has no contact as well.  So you now have a double whammy.

> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.34.136.92:
> 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836
> To: <sip:[email protected] 
> <mailto:sip%[email protected]> >;tag=rpz2agt27fobzvvd.o
> From: <sip:[email protected] <mailto:sip%[email protected]> 
> >;tag=55886164
> Call-ID: [email protected]
> CSeq: 1 INVITE
> Server: Sippy
> Content-Length: 0

On 06/07/2012 06:37 PM, Sven Evensen wrote: 

I totally agree! Just that often the answer is "Find another ITSP". In this 
case I want to put in some extra effort to resolve the issue. Not sure yet 
which side the issue originates. Sometimes it i easier to make a fix on our 
side than to get ITSP to change things. 

 

Sven

On Thu, Jun 7, 2012 at 11:31 AM, <[email protected]> wrote:

Sven Evensen <[email protected]> wrote on 07-06-2012 12:24:11:

> I saw that and I believe we made a patch once to add SDP to the 
> reInvite. Will have to check that. 
> 
> But the BYE from sipx says "sipXbridge;cause=205;text=\"Protocol 
> Error - 200 OK with no contact\"". I am trying to understand if sipX
> or Sippy is causing this. The original INVITE from Sippy has 
> "Anonymous" as contact. Also their ACK has no Contact (which is 
> legal they say) 
> 
> I remember Ranga mentioned a property in sipxBridge about relaxing 
> strict protocol. Is that still there? Is it worth trying? 
> 
> I really want to try to do something rather than saying the SIP 
> trunk is not compliant. Because they are not just a ITSP, but also a
> potential customer. 

Even ITSP's that are potential customers should at least be made aware that 
they don't adhere to standards. 
Always try to improve instead of engineer around issues. If it can't be fixed 
then that's another thing. 

Paul Scheepens 

> 
> Regards, 
> Sven

> On Thu, Jun 7, 2012 at 11:01 AM, Joegen Baclor <[email protected]> wrote: 
> sipX bridge disconnects because when it sent an INVITE without SDP 
> to Sippy B2BUA, that entity also responded with a 200 OK without 
> SDP.  That is uncompliance in Sippy part.  
> 
> 
> 
> INVITE sip:194.6.238.85:5061 SIP/2.0
> CSeq: 1 INVITE
> To: <sip:[email protected] 
> <mailto:sip%[email protected]> >;tag=rpz2agt27fobzvvd.o
> From: <sip:[email protected] <mailto:sip%[email protected]> 
> >;tag=55886164
> Call-ID: [email protected]
> Max-Forwards: 70
> Route: <sip:194.6.238.85;lr>
> User-Agent: sipXecs/4.4.0 sipXecs/sipxbridge (Linux)
> P-Asserted-Identity: <sip:[email protected] 
> <mailto:sip%[email protected]> >
> References: [email protected].
> 101-0;rel=chain;sipxecs-tag=request-refer-z9hg4bk-xx-4ab4jjiscj
> `j9ig27d8fw74puw
> Allow: INVITE,BYE,ACK,CANCEL,OPTIONS
> Supported: timer
> Via: SIP/2.0/UDP 176.34.136.92:
> 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836
> Contact: <sip:176.34.136.92:5080>
> Accept: application/sdp
> Content-Length: 0
> 
> -------------------------------------------------------------------------------------------------
> 
> SIP/2.0 200 OK
> Via: SIP/2.0/UDP 176.34.136.92:
> 5080;branch=z9hG4bK158560057fd8d521ed786e9511dbfcf0343836
> To: <sip:[email protected] 
> <mailto:sip%[email protected]> >;tag=rpz2agt27fobzvvd.o
> From: <sip:[email protected] <mailto:sip%[email protected]> 
> >;tag=55886164
> Call-ID: [email protected]
> CSeq: 1 INVITE
> Server: Sippy
> Content-Length: 0
> 
> 
> 
> On 06/07/2012 03:47 PM, Sven Evensen wrote: 
> Anyone?

> On Tue, Jun 5, 2012 at 1:05 PM, Sven Evensen <[email protected]> 
> wrote: 
> I have a new installation with a new SIP trunk provider. It seems 
> everything is working except transfer. sipx 4.4. 
> 
> Incoming external come through SIP trunk to attendant 
> Hit 1 to transfer to conference bridge 
> When transfer starts, call is dropped. 
> 
> This works find when a-party is internal. 
> 
> The problem might be in the INVITE coming from the REFER : 
> "SIP","Request: INVITE sip:194.6.238.85:5061, in-dialog" 
> causing them to answer with a 200 OK with no Contact header which 
> sipX does not like. 
> 
> Can anyone see from the attached pcap why this is happening 
> -- 
> Sven Evensen, Operations Consultant 
> OnRelay 
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-- 

Sven Evensen, Operations Consultant

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-- 

Sven Evensen, Operations Consultant

OnRelay

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This electronic message transmission contains information from OnRelay, Ltd., 
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-- 

Sven Evensen, Operations Consultant

OnRelay

Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │ 
mailto:[email protected] │  <http://www.onrelay.com/> www.onrelay.com


This electronic message transmission contains information from OnRelay, Ltd., 
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Registered in England No 04006093 ¦ Registered Office 1st Floor, 236 Gray's Inn 
Road, London WC1X 8HB

 

 





 

-- 

Sven Evensen, Operations Consultant

OnRelay

Elizabeth House │ 39 York Road, London SE1 7NQ, UK │ +44 (0) 207 902 8123 │ 
mailto:[email protected] │  <http://www.onrelay.com/> www.onrelay.com


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