This thread from SIPPING archive might help.
http://www1.ietf.org/mail-archive/web/sip/current/msg07289.html
Regards,
Gaurav
> -Original Message-
> From: [EMAIL PROTECTED] [mailto:sip-
> [EMAIL PROTECTED] On Behalf Of
> [EMAIL PROTECTED]
> Sent: Wednesday, June 27, 2007 11:12 AM
> To: Si
Thanks Robert and Bob. I missed the forking part.
I want to know whether such enhancements/defects in the RFCs are
tracked somewhere by the IETF?
The Record-Routing of mid-dialog requests by proxies is really
in-consistent, some does and others don't.
~Vikram
On 6/26/07, Robert Sparks <[EMAIL PRO
Hi All,
RFC 3312 discusses resource reservation usage with SIP. Can anyone
provide inputs on what are the possible resource reservation protocols
that can be used in such a manner?
Regards,
Varadaraj
The information contained in this electronic message and any attachments to
this message are
No, This means far end has put you on hold but u have your media in Send
only state.
Regards
Sunil verma
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of
varun
Sent: Wednesday, June 27, 2007 8:44 AM
To: [EMAIL PROTECTED]
Subject: [Sip-implementors] Ca
Hi,
If user B wants to put user A on Hold, it can send a =
sendonly to A and expects A to return a = recvonly.
This way there is a one way audio channel open from B
to A.
What if user A responds with a = inactive..is that
valid? Does it mean that media streams are on Hold in
both directions?
Thank
I too will apologize; I'm not sure if I'm overstepping bounds here.
I work at Valid8.com; we make conformance tests for (all? most?) VoIP protocols.
I'm on Sigtran and don't know the SIP side of things at all, but you could
contact [EMAIL PROTECTED] to find out more. Web site is http://www.valid8
[EMAIL PROTECTED] wrote:
> We (Pingtel) are running into an increasing number of customers who
> are having trouble with PSTN gateways. Specifically, they want to use
> more than one PSTN gateway, but it's difficult to have a proxy route
> calls reliably through multiple gateways, as there is no r
We (Pingtel) are running into an increasing number of customers who
are having trouble with PSTN gateways. Specifically, they want to use
more than one PSTN gateway, but it's difficult to have a proxy route
calls reliably through multiple gateways, as there is no reliable rule
to distinguish when
I apologize at the outset as this is not a protocol question but as most all
SIP implementors are on this list I am asking for help here.
I represent a small startup company and looking for SIP conformance testing
solution for our SIP Proxy and B2BUA based service broker.
We want to test protocol
Thanks Paul.
I had searched sip-implementors archive regarding 181.
I saw a discussion over to-tag in provisional responses.
Have shipping group reached a consensus over to-tag issue?
Could you let me know if you have any link which I can refer?
Thanks for your help.
Regards,
Sumin
On 6/26/07,
But be wary that lots of proxies don't Record-Route mid-dialog requests
(like they SHOULD), so things may go wrong.
More inline (to Vikram's response):
On Jun 26, 2007, at 10:05 AM, Bob Penfield wrote:
> Note that it is possible for the first NOTIFY request to be
> received before
> the 200-O
Note that it is possible for the first NOTIFY request to be received before
the 200-OK to the SUBSCRIBE and that the 'dialog' of the NOTIFY might be
different than the 'dialog' in 200-OK for the SUBSCRIBE if the SUBSCRIBE was
forked. In that case, the Record-Routes from the NOTIFY would be used.
The Record-Route set returned in the SUBSCRIBE 200 OK is considered
final for the dialog at UE's end.
The Record-Route set in the NOTIFY request received by the UE should
be discarded. This is also true for the Contact header coming in the
NOTIFY request.
Like re-INVITE, the Record-Route coming in
Dear all,
I come with a doubt regarding usage of the Record-Route header within a
SIP dialog established with SIP SUBSCRIBE. I would appreciate any
feedback you may have about this.
The use case is as follows:
- A User Agent SUBSCRIBEs to an event package and receives a final 200
OK response
BYE with either a different from or to tags
or better yet, use a none existent call-id
sarthakd wrote:
> Hi,
>
> I am trying to create an OUT OF DIALOG BYE.
>
> Logically, one of the options would be to change the from or to tags. Just
> wanted to know if I send a BYE without 'to' tag, will it b
Why would you want to generate an out-of-dialog BYE? It would be invalid
and useless.
Paul
sarthakd wrote:
> Hi,
>
> I am trying to create an OUT OF DIALOG BYE.
>
> Logically, one of the options would be to change the from or to tags. Just
> wanted to know if I send a BYE without 'to
> I am trying to create an OUT OF DIALOG BYE.
>
> Logically, one of the options would be to change the from or
> to tags. Just wanted to know if I send a BYE without 'to'
> tag, will it be an out-of-dialog bye?
Per rfc3261, yes. Per obsoleted rfc2543, maybe; it depends upon if tags
were used du
181 should follow the rules for any other provisional response. So you
could include SDP in the same cases where you might if it were a 183.
You are governed by the general offer/answer rules.
Paul
Sumin Seo wrote:
> Hi All,
>
> Let's say I want to play announcement to let user know th
Hi,
I am trying to create an OUT OF DIALOG BYE.
Logically, one of the options would be to change the from or to tags. Just
wanted to know if I send a BYE without 'to' tag, will it be an out-of-dialog
bye?
What are the other ways of simulating an out-of-dialog bye?
Thanks in Advance,
Sarthak
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