Thanks all.
-Original Message-
From: Attila Sipos [mailto:attila.si...@vegastream.com]
Sent: Tuesday, April 12, 2011 3:51 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] Phone-context header
Hi Manoj,
I did ask about phone-context
Dear All,
Please help me understand the use of phone-context field in the From
header of the INVITE. Is it globally significant? How to process the
INVITE coming to a proxy if the phone-context is not known to the proxy?
Spot-on documentation is highly appreciated
BR,
Manoj
Hi Nitin,
You can capture the RTP near the UAC and see whether the media is coming
up to the UAC after 183. If media is coming to the UAC, there might be a
setting in the UAC to enable playing RTP after receiving 183. I have
seen such configurations in some of the end points
Hope this helps
BR,
Thanks Brett
-Original Message-
From: Brett Tate [mailto:br...@broadsoft.com]
Sent: Tuesday, March 29, 2011 3:58 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: INVITE followed by 181
Some 181 and service examples can be found in rfc5359.
Some
Dear All,
How would the INVITE followed by 181 (Call being forwarded) looks like?
Should it contain the Diversion header? Any documentation is highly
appreciated
Thanks
BR,
Manoj
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What happens if there's a SIP call processing issue in the call server
(UAS) while having perfect network connectivity between UAC and UAS? So
100-Trying doesn't have any deal whether the transport protocol is TCP
or UDP, Signaling flow must not change depending on the transport
protocol
BR,
Manoj
Thanks Brett. Appreciate the support
-Original Message-
From: Brett Tate [mailto:br...@broadsoft.com]
Sent: Monday, February 21, 2011 6:16 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: Proxy-Authorization
> What is the exact format of the Pr
Dear All,
What is the exact format of the Proxy-Authorization header in the INVITE
after the challenging 407 of the SIP Proxy? This is not specified in the
RFC2361. Please help me to find a reference
Thanks
BR,
Manoj
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Thanks a lot
-Original Message-
From: Schwarz, Albrecht (Albrecht)
[mailto:albrecht.schw...@alcatel-lucent.com]
Sent: Friday, January 21, 2011 1:17 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Cc: tec...@sipforum.org
Subject: RE: Re-transmission in T.38
This is
Dear All,
Please help me to understand the packet repetition in T.38.
1. How many packets are supposed to be repeated and is the number of
repeated packets are different from different fax signal to fax signal?
(such as V21-preamable, cng etc)
2. Is this packet repetition is only relevant to
Dear All,
Apologies if I'm asking something out of the scope of this forum.
However I believe there are IMS experts who can advice me on the
following query.
As per the standard, where should we define the call-barring information
(per user) in IMS, is it in the HSS or the telephony server?
Than
True. This is a common call scenario, especially when passing few
network elements
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Vivek Batra
Sent: Monday, October 11, 2010 10:13 AM
To: 'Ayyanar
Hi vijay,
Yes! As an example 183-session progress followed by 180-rining
Br, manoj
--- original message ---
From: "Vijay S Nair"
Subject: [Sip-implementors] UAS behavior : Multiple 18x messages
Date: 9th October 2010
Time: 8:59:08 pm
Hi All,
Here is the scenario,
UA1
Thanks!
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
$...@r\/|>r!`/@
Sent: Wednesday, October 06, 2010 12:05 PM
To: Alex Balashov; Manoj Priyankara [TG]
Cc: sip-implement
] Detection of Cable disconnections in SIP
On 10/06/2010 01:52 AM, Manoj Priyankara [TG] wrote:
> If there's a LAN cable disconnect during a SIP session that is during
a
> call how does the other party recognize it? As an example, let us
> consider UA A and UA B are in a call. Suddenly UA
Hi All,
If there's a LAN cable disconnect during a SIP session that is during a
call how does the other party recognize it? As an example, let us
consider UA A and UA B are in a call. Suddenly UA A disconnects. How
does the UA B know A is no longer connected?
Thanks
BR,
Manoj
plays back the music till
the call is transferred to the relevant party.
Thanks for the comments!
Manoj
-Original Message-
From: zabi [mailto:mohamed.zabiu...@globaledgesoft.com]
Sent: Thursday, August 19, 2010 12:13 PM
To: Avasarala Ranjit-A20990
Cc: Alok 2 Tiwari; Manoj Priyankara [TG
Thanls Raju.
-Original Message-
From: P A Raju [mailto:par...@hyd.hellosoft.com]
Sent: Thursday, August 19, 2010 12:15 PM
To: Manoj Priyankara [TG]; sip-implementors
Subject: Re: [Sip-implementors] Invite without SDP
Dear Manoj,
See my comments inline.
Thanks & Regards,
Dear All,
If a particular UA sends and INVITE to hold a call, do we have to
include SDP?
If SDP is not included in the INVITE, Should the other party include SDP
in OK message?
Thanks
BR,
Manoj
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Hi,
Please check whether the called party supports the codec g.728
BR,
Manoj
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Franz Edler
Sent: Wednesday, May 05, 2010 5:53 PM
To: sip-implementors
Thanks Satya
-Original Message-
From: Satyakumar [mailto:satyam_...@hyd.hellosoft.com]
Sent: Wednesday, March 24, 2010 11:05 AM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] different use of SIP INVITE message
Hi,
The SDP is
Dear All,
What is the difference between the INVITE message used for call features (such
as Call hold, transfer etc) and the INVITE message used for codec negotiation (
such as T.38 fax relay) ?
Thanks
BR,
Manoj
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Sip
Hi Palie,
Thanks for the information.
BR,
Manoj
-Original Message-
From: Kamalakanta Palei (kpalei) [mailto:kpa...@cisco.com]
Sent: Thursday, December 10, 2009 8:06 AM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu; Rajani
Subject: RE: [Sip-implementors
-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Tomasz Zieleniewski
Sent: Thursday, December 10, 2009 12:57 AM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] differences between Via,Record-Route and
Route headers
Hi Manoj,
Some time ago I
Thanks Alok!
-Original Message-
From: Alok 2 Tiwari [mailto:alok2.tiw...@aricent.com]
Sent: Wednesday, December 09, 2009 11:46 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] differences between Via, Record-Route
and Route headers
Hi
Dear All,
Can anyone explain the differences between Via, Record-Route and Route headers?
Further, are there any other routing related Headers associated with SIP?
Thanks!
BR,
Manoj
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Thanks Paulo!
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Paulo Borelli
Sent: Friday, November 27, 2009 4:52 AM
To: Manoj Priyankara [TG]; sunil.bha...@wipro.com;
sip-implementors
Dear All,
Can anyone explain how ptime affects in call setups?
Should all the UA's include ptime in the SDP of the INVITE?
If it is included, and does not match with the packet time of the other
party what would happen?
Little information could be found in RFC 2327. Pls help
Thanks
BR,
Manoj
Dear All,
Got few questions as follows regarding the Codec negotiation in SDP
1. Can there be an INVITE without SDP? (Resulting no Codec offer to the
other party)
2. If so, what would be the response from the other party (Should it
reply with any codec supported, or all the capable codec's)
3. C
Dear All,
How we can find out the best value of the jitter buffer for a VoIP end
point? Any recommendations for the same?
Thanks
BR,
Manoj
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Thanks Abhishek.
BR,
Manoj
-Original Message-
From: Abhishek Dhammawat [mailto:abhishek.dhamma...@aricent.com]
Sent: Friday, October 23, 2009 12:00 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: T.38 Specifications
Hi Manoj
Please find below some
Thanks Albrecht! Will try to download the latest release.
BR,
Manoj
-Original Message-
From: Schwarz Albrecht [mailto:albrecht.schw...@alcatel-lucent.de]
Sent: Friday, October 23, 2009 12:15 PM
To: Abhishek Dhammawat; Manoj Priyankara [TG];
sip-implementors@lists.cs.columbia.edu
Subject
Dear All,
I need to find the standards / call scenarios related to SIP-T.38 fax.
In the ITU, I could find only H.323. Please help
Thanks
BR,
Manoj
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OK Thanks Sir
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Bharat S
Sent: Monday, September 21, 2009 3:18 PM
To: Manoj Priyankara [TG]; Avasarala Ranjit-A20990;
sip-implementors
Hi Ranjit,
Thanks!
In the SDP, which field we have the media IP? ( is it c= IN . ?)
BR,
Manoj
-Original Message-
From: Avasarala Ranjit-A20990 [mailto:ran...@motorola.com]
Sent: Monday, September 21, 2009 1:02 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Dear All,
In which SIP message we have the IP address to which media will be
pointed?
Thanks
BR,
Manoj
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Hi Keerthi,
Thanks!
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Keerthi
Sent: Wednesday, August 19, 2009 10:43 AM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu
Subject
Dear All,
When an INVITE passes through a SIP proxy, should the contact header
change?
Thanks
BR,
Manoj
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pkyzi...@cisco.com]
> Sent: Sun 8/9/2009 2:48 AM
> To: Manoj Priyankara [TG]
> Cc: sip-implementors@lists.cs.columbia.edu
> Subject: Re: [Sip-implementors] SIP OPTIONS
>
>
>
> Manoj Priyankara [TG] wrote:
> > Dear All,
> >
> > According to the RFC 3261, SI
to see the availability of the UAS.
BR,
Manoj
-Original Message-
From: Paul Kyzivat [mailto:pkyzi...@cisco.com]
Sent: Sun 8/9/2009 2:48 AM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] SIP OPTIONS
Manoj Priyankara [TG] wrote:
Thanks Sir :)
-Original Message-
From: Abhishek Dhammawat [mailto:abhishek.dhamma...@aricent.com]
Sent: Thursday, August 06, 2009 7:25 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: SIP OPTIONS
Hi
In my opinion OPTIONS should be used for querying the
Dear All,
According to the RFC 3261, SIP OPRIONS message should be used to query
the statue of other UAC or the UAS. Is it OK to use the OPTIONS as a
keep alive message to know whether the UAS is alive?
Is it necessary to send the OPTIONS message from a registered user or is
it possible to send t
Hi,
Did you try eye-beam from counter path?
BR,
Manoj
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Sudhir Kumar Reddy
Sent: Thursday, July 16, 2009 3:43 PM
To: sip-implementors@lists.cs.columbi
Dear all,
What are the differences between a SIP Proxy and an OUTBOUND SIP Proxy ?
Thanks
BR,
Manoj
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Thanks sir !
From: Abhinesh Patil [mailto:abhinesh.pa...@rancoretech.com]
Sent: Tuesday, July 14, 2009 11:16 AM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] codec negotiation
You can read - RFC 3264 an
Dear All,
Can somebody explain me how codec negotiation takes place in SIP?
BR,
Manoj
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Thanks Ranga...
BR,
Manoj
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of M.
Ranganathan
Sent: Wednesday, July 01, 2009 8:33 PM
To: Manoj Priyankara [TG]
Cc: Paul Kyzivat; sip-implementors
Thanks Paul
Go the idea, apologies for making it bit complicated,
BR,
Manoj
-Original Message-
From: Paul Kyzivat [mailto:pkyzi...@cisco.com]
Sent: Wednesday, July 01, 2009 9:04 PM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors
connect the call
legs from the SS and release the IP-PBX from the picture?
Thanks for the expert support
BR,
Manoj
-Original Message-
From: Rastogi, Vipul (Vipul) [mailto:vrast...@avaya.com]
Sent: Wednesday, July 01, 2009 2:45 PM
To: Manoj Priyankara [TG]; Paul Kyzivat
Cc: sip-implementors
: Tuesday, June 30, 2009 6:50 PM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] Call Transfer through a SIP trunk
Manoj,
Virtually none of the terms you use below have well defined meanings in
sip standards. Many people use such terms, but AFAIK
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
if the B party is a legacy subscriber connected via a
H.248 or MGCP media gateway ?
Cheers!
Manoj
-Original Message-
From: Dale Worley [mailto:dwor...@nortel.com]
Sent: Fri 6/12/2009 9:43 PM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip
Thanks Victor
//
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Victor
Pascual Ávila
Sent: Thursday, June 11, 2009 3:18 PM
To: Manoj Priyankara [TG]
Cc: sip-implementors@lists.cs.columbia.edu
Dear All,
Could one of you please explain me how to handle following situation in
a SIP session?
Let us imagine that UAC's A and B are in a call and due to a network
connectivity problem, user A disconnects without sending any message to
the UAS.
Then the UAS still thinks that UAC A is alive. Of
Hi all,
Apologies...
Thanks
BR,
Manoj
-Original Message-
From: Iñaki Baz Castillo [mailto:i...@aliax.net]
Sent: Wednesday, April 29, 2009 12:28 PM
To: sip-implementors@lists.cs.columbia.edu; Manoj Priyankara [TG]
Subject: Re: [Sip-implementors] P-CSCF Security functions
El Miércoles
Hi All,
I have seen some equipment vendors have implement security functions
such as DDoS protection in the P-CSCF. According to my understanding
this should be a function of the SBC. Is this a standard thing?
Thanks
BR,
Manoj
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Hi All,
I have seen some equipment vendors have implement security functions
such as DDoS protection in the P-CSCF. According to my understanding
this should be a function of the SBC. Is this a standard thing?
Thanks
BR,
Manoj
-Original Message-
From: sip-implementors-boun...@lists.cs.
Please try OpenSIPS
BR,
Manoj
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
nadeige Da costa
Sent: Monday, April 06, 2009 11:58 AM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-imple
Thanks Guys!
BR,
Manoj
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Alejandro Orellana
Sent: Monday, March 23, 2009 6:12 PM
To: Manoj Priyankara [TG]; sip-implementors@lists.cs.columbia.edu
Hi All,
Can anyone tell me the exact call flow associated with Blind Call
Transfer of SIP ?
BR,
Manoj
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Hi All,
Is there any standard limiting the number of registrations from same
IP:port for a P-CSCF ?
BR,
Manoj
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