Re: [Sip-implementors] PRACK-Scenario

2009-06-30 Thread Arunachala
Hi, In the first scenario, the UAS is NOT behaving according to RFC3262. It MUST reject the INVITE with 420 (Bad Extension) if it does NOT support 100rel. If you really want to handle this scenario, I would suggest silently dropping the non-reliable 180, as you were expecting a reliable 180.

[Sip-implementors] Guideline to display connected identity on the UE

2009-06-30 Thread Govind, Mahesh (NSN - IN/Bangalore)
HI , What is the guideline to display "connected identity " in the following cases 1) P-Asserted-Identity and changed From Header is there in a request Do we display P-Asserted Identity or user part of "From Header"? I feel "From Header" user part has to be displayed because it is co

Re: [Sip-implementors] PRACK-Scenario

2009-06-30 Thread soma bhargava
Hi, Thanks for the reply, but rfc 3262 does not clearly mention what to do in these scenarios. Regards, Soma On Tue, 2009-06-30 at 12:43 -0400, Alejandro Orellana wrote: > Please refer to http://www.ietf.org/rfc/rfc3262.txt > > thanks > > On Tue, Jun 30, 2009 at 11:48 AM, soma bhargava > wrot

Re: [Sip-implementors] Call Transfer through a SIP trunk

2009-06-30 Thread Manoj Priyankara [TG]
Hi Paul, Sorry for using more generic terms. In this case the IP-PBX acts as a B2BUA and the SS acts as a Proxy BR, Manoj -Original Message- From: sip-implementors-boun...@lists.cs.columbia.edu [mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Paul Kyzivat Sent: Tuesda

Re: [Sip-implementors] About Transport Selection in Response

2009-06-30 Thread Dale Worley
A major difficulty is that the only IP/port/transport that you *know* the sender is listening on is the one specified in the Via. Many if not most SIP elements will properly process a SIP message no matter what port it is received on, but there is no guarantee that the sending element is listening

Re: [Sip-implementors] PRACK-Scenario

2009-06-30 Thread Alejandro Orellana
Please refer to http://www.ietf.org/rfc/rfc3262.txt thanks On Tue, Jun 30, 2009 at 11:48 AM, soma bhargava < soma.bharg...@globaledgesoft.com> wrote: > Hi All, > > What is the behaviour of UAC in following scenario: > > 1. UAC sends INVITE with 100rel in both requires and supported headers. > UA

[Sip-implementors] PRACK-Scenario

2009-06-30 Thread soma bhargava
Hi All, What is the behaviour of UAC in following scenario: 1. UAC sends INVITE with 100rel in both requires and supported headers. UAS sends 180 prov resp without 100rel in both requires and supported headers. what should the UAC do? 2. Invite is sent without 100rel in both require and support

Re: [Sip-implementors] Call Transfer through a SIP trunk

2009-06-30 Thread Paul Kyzivat
Manoj, Virtually none of the terms you use below have well defined meanings in sip standards. Many people use such terms, but AFAIK there are not well understood and consistent implementations for them. So it is not possible to answer your question as posed. Tell us whether your SS and IP-PBX

Re: [Sip-implementors] About Transport Selection in Response

2009-06-30 Thread Michael Procter
You may also find RFC 4483 (Content-Indirection in SIP) useful, depending on support in the endpoints. Michael 2009/6/30 Brett Tate : > Yes; the RFCs currently require the same transport be used. > > Some vendors will likely accommodate what you are attempting.  However just > because the UAC/p

Re: [Sip-implementors] About Transport Selection in Response

2009-06-30 Thread Brett Tate
Yes; the RFCs currently require the same transport be used. Some vendors will likely accommodate what you are attempting. However just because the UAC/proxy might be willing to receive the response doesn't mean they will actually process it as you are hoping; thus it might it might be hard to

[Sip-implementors] Call Transfer through a SIP trunk

2009-06-30 Thread Manoj Priyankara [TG]
Dear all, Please consider the following scenario. Assume that there is a standard Soft Switch; and an IP PBX is connected to the SS using a SIP trunk. IP PBX has a user (Say A) An incoming call comes from an external party through the SS to user A. User A transfers that call to an external par