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Today's Topics:
1. retrying servers in the SRV response:
draft-ietf-sip-srv-06.txt (Manpreet Singh)
2. Re: how to transfer a call use sip (Broussard, Dan)
3. Re: [Sip-implementors] retrying servers in the SRV
response:draft -ietf-sip-srv-06.txt (Manpreet Singh)
4. Re: how to transfer a call use sip (Srinivas.Pratapa)
------------------------------------------------------------
----------
Message: 1
Date: Fri, 4 Aug 2006 16:56:47 -0400
From: Manpreet Singh <msingh ibasis.net>
Subject: [SIPForum-discussion] retrying servers in the SRV
response:
draft-ietf-sip-srv-06.txt
To: discussion sipforum.org
Cc: sip-implementors cs.columbia.edu
Message-ID:
<6E9C6C7D27E3714190801A551BE37610030CF257 server507.bur.ibasis.net>
Content-Type: text/plain; charset="us-ascii"
Hi
I read the draft for locating SIP servers using SRV and
there is a section
on how to retry in the event of failure. The only thing
mentioned there is
if the client gets a 503 or there are transport level
timeouts. Now is that
the only implementation? Can the client implement retries
based on certain
4xx or 5xx responses? I was testing a case where I was
sending 500 and the
client wont retry but for 503 it would. So is this really
the final
implementation or its really client specific.
Thanks
Manpreet
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Message: 2
Date: Fri, 4 Aug 2006 15:10:36 -0600
From: "Broussard, Dan" <Dan.Broussard Level3.com>
Subject: Re: [SIPForum-discussion] how to transfer a call
use sip
To: "Gunnar Hellstrom" <gunnar.hellstrom omnitor.se>, "Sreedhar
Pampati" <Sreedhar_Pampati net.com>, "liurf" <liu.renfeng gmail.com>,
<discussion sipforum.org>
Message-ID:
<25347E904A7464489BD59089E714E6180B082799 idc1exc0006.corp.global.level3.com>
Content-Type: text/plain; charset="us-ascii"
To handle a transfer in our network we usually see an
re-invite..
Liurf...if you want a sip ladder let me know I will put one
together for
you.
Dan Broussard
Level 3 Communications
________________________________
From: discussion-bounces sipforum.org
[mailto:discussion-bounces sipforum.org] On Behalf Of
Gunnar Hellstrom
Sent: Friday, August 04, 2006 4:39 PM
To: Sreedhar Pampati; liurf; discussion sipforum.org
Subject: Re: [SIPForum-discussion] how to transfer a call
use sip
And do not forget that REFER is just a component in the
process
described in
http://www.ietf.org/internet-drafts/draf
t-ietf-sipping-cc-transfer-06.tx
t
Regards
Gunnar
-----Original Message-----
From: discussion-bounces sipforum.org
[mailto:discussion-bounces sipforum.org]On Behalf Of
Sreedhar Pampati
Sent: Friday, August 04, 2006 5:45 PM
To: liurf; discussion sipforum.org
Subject: Re: [SIPForum-discussion] how to transfer a call
use
sip
Best place to start would be the following draft that lists
the
call flow and example message contents:
http://ietf.org/internet-drafts/draft-ie
tf-sipping-service-examples-10.t
xt
Then refer the RFC 3515 (REFER method)
HTH
Sreedhar
-----Original Message-----
From: discussion-bounces sipforum.org
[mailto:discussion-bounces sipforum.org] On Behalf Of
liurf
Sent: Thursday, August 03, 2006 7:20 PM
To: discussion sipforum.org
Subject: [SIPForum-discussion] how to transfer a call use
sip
Dear Sir or Madam:
Can anybody tell me how to use sip to transfer a call. I
want to
know the detail of how to transfer a call under sip.
Thanks very much
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Message: 3
Date: Fri, 4 Aug 2006 18:28:00 -0400
From: Manpreet Singh <msingh ibasis.net>
Subject: Re: [SIPForum-discussion] [Sip-implementors]
retrying servers
in the SRV response:draft -ietf-sip-srv-06.txt
To: "Sanjay Sinha (sanjsinh)" <sanjsinh cisco.com>,
discussion sipforum.org
Cc: sip-implementors cs.columbia.edu
Message-ID:
<6E9C6C7D27E3714190801A551BE37610030CF271 server507.bur.ibasis.net>
Content-Type: text/plain; charset="us-ascii"
Ok, so any 5xx, the client should try right? It simply said
503 so I was
wondering. Also the retry mechanism using SXRV for SIP thing
is really
handled at the stack level and not the application level
right?
Manpreet
-----Original Message-----
From: Sanjay Sinha (sanjsinh) [mailto:sanjsinh cisco.com]
Sent: Friday, August 04, 2006 6:30 PM
To: Manpreet Singh; discussion sipforum.org
Cc: sip-implementors cs.columbia.edu
Subject: RE: [Sip-implementors] retrying servers in the SRV
response:draft-ietf-sip-srv-06.txt
This draft is RFC 3263.
Also I think the client would try other hosts for error
responses unless
error response indicates a global failure, like 6XX.
Sanjay
-----Original Message-----
From: sip-implementors-bounces cs.columbia.edu
[mailto:sip-implementors-bounces cs.columbia.edu] On Behalf
Of Manpreet
Singh
Sent: Friday, August 04, 2006 4:57 PM
To: discussion sipforum.org
Cc: sip-implementors cs.columbia.edu
Subject: [Sip-implementors] retrying servers in the SRV
response:draft-ietf-sip-srv-06.txt
Importance: High
Hi
I read the draft for locating SIP servers using SRV and
there is a section
on how to retry in the event of failure. The only thing
mentioned there is
if the client gets a 503 or there are transport level
timeouts. Now is that
the only implementation? Can the client implement retries
based on certain
4xx or 5xx responses? I was testing a case where I was
sending 500 and the
client wont retry but for 503 it would.
So is this really the final implementation or its really
client specific.
Thanks
Manpreet
_______________________________________________
Sip-implementors mailing list
Sip-implementors cs.columbia.edu
https://lists.cs.columbia.edu/cucslists/listinf
o/sip-implementors
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Message: 4
Date: Tue, 8 Aug 2006 12:58:29 -0400
From: "Srinivas.Pratapa"
<Srinivas.Pratapa resolvity.com>
Subject: Re: [SIPForum-discussion] how to transfer a call
use sip
To: "Broussard, Dan" <Dan.Broussard level3.com>, "Gunnar Hellstrom"
<gunnar.hellstrom omnitor.se>, "Sreedhar
Pampati"
<Sreedhar_Pampati net.com>, "liurf"
<liu.renfeng gmail.com>,
discussion sipforum.org
Message-ID:
<DA317F24760CC74EA3151DE3AAC174EF02C2A8DF MI8NYCMAIL05.Mi8.com>
Content-Type: text/plain; charset="us-ascii"
Hey Dan,
We've been trying to get transfer working over SIP. We are
using
Bandwidth.com as our provider which in turn uses level3 for
their
infrastructure. Do you happen to have any documentation on
how the
transfer request should be sent? We are trying this transfer
using VXML
constructs and none of the approaches (using
consult/blind/bridge) seem
to work.
thanks
Srinivas
________________________________
From: discussion-bounces sipforum.org
[mailto:discussion-bounces sipforum.org] On Behalf Of
Broussard, Dan
Sent: Friday, August 04, 2006 4:11 PM
To: Gunnar Hellstrom; Sreedhar Pampati; liurf;
discussion sipforum.org
Subject: Re: [SIPForum-discussion] how to transfer a call
use sip
To handle a transfer in our network we usually see an
re-invite..
Liurf...if you want a sip ladder let me know I will put one
together for
you.
Dan Broussard
Level 3 Communications
________________________________
From: discussion-bounces sipforum.org
[mailto:discussion-bounces sipforum.org] On Behalf Of
Gunnar Hellstrom
Sent: Friday, August 04, 2006 4:39 PM
To: Sreedhar Pampati; liurf; discussion sipforum.org
Subject: Re: [SIPForum-discussion] how to transfer a call
use sip
And do not forget that REFER is just a component in the
process
described in
http://www.ietf.org/internet-drafts/draf
t-ietf-sipping-cc-transfer-06.tx
t
Regards
Gunnar
-----Original Message-----
From: discussion-bounces sipforum.org
[mailto:discussion-bounces sipforum.org]On Behalf Of
Sreedhar Pampati
Sent: Friday, August 04, 2006 5:45 PM
To: liurf; discussion sipforum.org
Subject: Re: [SIPForum-discussion] how to transfer a call
use
sip
Best place to start would be the following draft that lists
the
call flow and example message contents:
http://ietf.org/internet-drafts/draft-ie
tf-sipping-service-examples-10.t
xt
Then refer the RFC 3515 (REFER method)
HTH
Sreedhar
-----Original Message-----
From: discussion-bounces sipforum.org
[mailto:discussion-bounces sipforum.org] On Behalf Of
liurf
Sent: Thursday, August 03, 2006 7:20 PM
To: discussion sipforum.org
Subject: [SIPForum-discussion] how to transfer a call use
sip
Dear Sir or Madam:
Can anybody tell me how to use sip to transfer a call. I
want to
know the detail of how to transfer a call under sip.
Thanks very much
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