Re: [Sip-implementors] Regarding Subscription Termination

2011-07-05 Thread Amith R R
Ok, I got it. Thanks for your support. Thanks and Regards, Amith -Original Message- From: Brett Tate [mailto:br...@broadsoft.com] Sent: Tuesday, July 05, 2011 5:05 PM To: Amith R R; Sip-implementors@lists.cs.columbia.edu Subject: RE: [Sip-implementors] Regarding Subscription Termination

Re: [Sip-implementors] Contact in a 200 OK

2011-07-05 Thread Worley, Dale R (Dale)
From: sip-implementors-boun...@lists.cs.columbia.edu [sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Simon Gregory [si...@myphones.com] If a UA sends a REGISTER with say a@b.c: as a contact (binding) would it ever be conceivable that a r

[Sip-implementors] test

2011-07-05 Thread Paul Kyzivat
please ignore ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors

Re: [Sip-implementors] Contact in a 200 OK

2011-07-05 Thread Brett Tate
> Has anyone ever seen the contact of a 200 OK return URI > info about the proxy / registrar itself? > i.e. contact in 200 OK = a@regip:regport Yes; it can happen if SBC changes Contact within REGISTER before sending to registrar and subsequently did not change it back within the 200 response(s

Re: [Sip-implementors] Contact in a 200 OK

2011-07-05 Thread Iñaki Baz Castillo
2011/7/5 Simon Gregory : > > I was under the impression that RFC3261 Sect 10.3, subsect 8 makes it > explicit that a proxy / registrar should return (in the contact) the > bindings (typically 1 in my simple scenario) that is basically a reflection > of the contact sent in the original REGISTER. Ye

[Sip-implementors] Contact in a 200 OK

2011-07-05 Thread Simon Gregory
If a UA sends a REGISTER with say a@b.c: as a contact (binding) would it ever be conceivable that a registrar would return anything other than that binding in a 200 OK (I am assuming here that this UA is only registering once). More specifically, Has anyone ever seen the contact of a 2

Re: [Sip-implementors] Re-Invite codec renegotiation.

2011-07-05 Thread Johnson, Michael A
Thanks Gents, I'm sorry my terminology has caused confusion. However The responses received have clarified things for me. Seems the vendor is not in violation of the standard, just not implementing the solution well. I'll try another tack to get the way they handle the codec negotiation modifie

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Brez Borland
On Tue, Jul 5, 2011 at 1:32 PM, Iñaki Baz Castillo wrote: > 2011/7/5 Brez Borland : > > I think this is a complete misinterpretation of the nature of > Record-Route > > mechanism. As opposed to RURI, Record-Route gives a one-hop behavior, and > > indicating SIPS scheme, without transport paramete

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Iñaki Baz Castillo
2011/7/5 Brez Borland : > I think this is a complete misinterpretation of the nature of Record-Route > mechanism. As opposed to RURI, Record-Route gives a one-hop behavior, and > indicating SIPS scheme, without transport parameter, is enough. Right. -- Iñaki Baz Castillo __

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Brez Borland
On Tue, Jul 5, 2011 at 12:58 PM, Iñaki Baz Castillo wrote: > 2011/7/5 Brez Borland : > > Have a look at: > > pjsip/src/pjsua-lib/pjsua_call.c: get_secure_level(); > > > > It seems to me that PJSIP does not support SIPS scheme in Record-Route. > That > > said, you have to put SIP and "transport=tl

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Brez Borland
On Tue, Jul 5, 2011 at 1:14 PM, Brez Borland wrote: > On Tue, Jul 5, 2011 at 1:01 PM, Iñaki Baz Castillo wrote: > >> 2011/7/5 Iñaki Baz Castillo : >> > 2011/7/5 Brez Borland : >> >> A very interesting thing I have just encountered, rfc5603 Section 3.1.4 >> >> says: >> >> >> >> The transport=tls

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Brez Borland
On Tue, Jul 5, 2011 at 1:01 PM, Iñaki Baz Castillo wrote: > 2011/7/5 Iñaki Baz Castillo : > > 2011/7/5 Brez Borland : > >> A very interesting thing I have just encountered, rfc5603 Section 3.1.4 > >> says: > >> > >> The transport=tls parameter has never been defined in an RFC, but only > in > >>

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Iñaki Baz Castillo
2011/7/5 Iñaki Baz Castillo : > But note that my case is a bit special as the initial INVITE arrives > via TLS to the proxy, but does not contain a sips schema no where. So > maybe in this case (and just in this case) the URI in the Record-Route > should not contain a sips: schema and instead conta

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Iñaki Baz Castillo
2011/7/5 Iñaki Baz Castillo : > 2011/7/5 Brez Borland : >> A very interesting thing I have just encountered, rfc5603 Section 3.1.4 >> says: >> >> The transport=tls parameter has never been defined in an RFC, but only in >> some of the Internet drafts between [RFC2543] and [RFC3261]. >> >> So they c

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Iñaki Baz Castillo
2011/7/5 Brez Borland : > A very interesting thing I have just encountered, rfc5603 Section 3.1.4 > says: > > The transport=tls parameter has never been defined in an RFC, but only in > some of the Internet drafts between [RFC2543] and [RFC3261]. > > So they couldn't really deprecate it, or restric

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Iñaki Baz Castillo
2011/7/5 Brez Borland : > Have a look at: > pjsip/src/pjsua-lib/pjsua_call.c: get_secure_level(); > > It seems to me that PJSIP does not support SIPS scheme in Record-Route. That > said, you have to put SIP and "transport=tls" when writing Record-Route. So UGLY Thanks for pointing it out. --

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Brez Borland
A very interesting thing I have just encountered, rfc5603 Section 3.1.4 says: The transport=tls parameter has never been defined in an RFC, but only in some of the Internet drafts between [RFC2543] and [RFC3261]. So they couldn't really deprecate it, or restrict it, cause it never really existed.

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Brez Borland
Have a look at: pjsip/src/pjsua-lib/pjsua_call.c: get_secure_level(); It seems to me that PJSIP does not support SIPS scheme in Record-Route. That said, you have to put SIP and "transport=tls" when writing Record-Route. Regards, Brez On Tue, Jul 5, 2011 at 12:16 PM, Brez Borland wrote: > Hi

Re: [Sip-implementors] Regarding Subscription Termination

2011-07-05 Thread Brett Tate
> As per the below description, AS has to send NOTIFY > and Core network has to reject it in order to cancel > the subscription. Those are not the only ways to terminate the subscription. If the NOTIFY is to terminate the subscription, the transferor does not have to reject the NOTIFY. I

Re: [Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Brez Borland
Hi Inaki, On Tue, Jul 5, 2011 at 10:43 AM, Iñaki Baz Castillo wrote: > Hi, I'm experimenting a problem with a client (PJSIP) connecting to a > proxy via TLS: > > - The client uses "sip:" scheme in INVITE headers and "sip:" with > ";transport=tls" in Contact header. It is valid according to some

[Sip-implementors] INVITE over TLS without "sips". Which URI must add the proxy in Record-Route?

2011-07-05 Thread Iñaki Baz Castillo
Hi, I'm experimenting a problem with a client (PJSIP) connecting to a proxy via TLS: - The client uses "sip:" scheme in INVITE headers and "sip:" with ";transport=tls" in Contact header. It is valid according to some RFC. - The proxy routes the INVITE via UDP and adds a Record-Route like this: