Hello,
I think it is not clear.
Do have forking ?
Do want have multiple contacts.
Regards!
Valdemar
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
budi wibowo
Sent: Tuesday, March 27, 201
Beatty explanation , and *you* will not pay the call for *caller*.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Paul Kyzivat
Sent: Thursday, January 12, 2012 11:08 AM
To: sip-implementors@l
Thanks!
From: ext Sumit Jindal [mailto:sumitjindal.n...@gmail.com]
Sent: Thursday, November 24, 2011 1:09 AM
To: Aman Aggarwal
Cc: Pavesi, Valdemar (NSN - US/Irving);
sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] UPDATE delay to send 200ok final
response
Hi
!
Valdemar
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of
Pavesi, Valdemar (NSN - US/Irving)
Sent: Wednesday, November 30, 2011 5:14 PM
To: ext Nitin Kapoor
Cc: sip-implement...@cs.columbia.edu;
sip
quot;
Regards!
Valdemar
From: ext Nitin Kapoor [mailto:nitinkapo...@gmail.com]
Sent: Wednesday, November 30, 2011 2:56 PM
To: Pavesi, Valdemar (NSN - US/Irving)
Cc: sip-implement...@cs.columbia.edu;
sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] 488 Not Accep
Hello Nitin Kapoor,
Do you have the first OFFER and ANSWER ?
The first offer CODECS: 4 18 101 , we have to see the ANSWER , at
moment looks ok , the session-ID is "o=Sonus_UAC 16515 29422"
v=0
o=Sonus_UAC 16515 29422 IN IP4 209.58.46.49
s=SIP Media Capabilities
c=IN IP4 209
Hello Nitin Kapoor,
Do you have the first OFFER and ANSWER ?
The first offer CODECS: 4 18 101 , we have to see the ANSWER , at
moment looks ok , the session-ID is "o=Sonus_UAC 16515 29422"
v=0
o=Sonus_UAC 16515 29422 IN IP4 209.58.46.49
s=SIP Media Capabilities
c=IN IP4 209
Hello ,
We have to send update to change the session :
a) We want it
1 50.987832 10.48.4.2 10.52.39.194 SIP/SDP
Request: UPDATE sip:10.52.39.194:5060;transport=UDP, with session
description
2 50.987832 10.52.39.194 10.48.4.2 SIP/SDP
Status
Hello,
For Session Timers in the Session Initiation Protocol (SIP). re-INVITE
due Session-Expires must keep the Max-Forwards from original INVITE
?
I am not able to find it clear on RFC.
++
INVITE sip:15109794...@volte.com SIP/2.0
Content-Length:244
From:;tag=Z5diY7j
It will be registered by contact header not by to-header
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Olle E. Johansson
Sent: Monday, October 31, 2011 7:20 AM
To:
sip-implement...@lists.cs.
Hello Vineet Menon,
The 100-trying will be used to stop the timer T1 (500MSEC) , If any
subsequent response 18x will take longer than 500msec than 100trying
will
Be send back to stop the timer T1. Check the timestamp when 180ringing
was send.
Regards!
Valdemar
-Original Message-
F
Nancy,
No , the blank line must be only before the content, in your case
"Hello World".
Regards!
Valdemar
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Nancy Greene
Sent: Friday, Septembe
http://www.tech-invite.com/Ti-sip-service-10.html
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Anand A
Sent: Friday, September 16, 2011 7:42 AM
To: sip-implementors@lists.cs.columbia.edu
Sub
Looks like a-side don't want talk.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Sproul, Barry K
Sent: Thursday, September 15, 2011 1:56 PM
To: sip-implementors@lists.cs.columbia.edu
Subject
Is the username/password correct ? if header Authorization is not
correct the server will keep sending the challenge up to a limit
(like 50 times)
To avoid loop. After reach the maximum you must see 403 instead 401.
-Original Message-
From: sip-implementors-boun...@lists.cs.co
Hello,
I want just give a hit how we had solved it a long time ago.
a) the switch will send INVITE without SDP to BOB_ASIDE
b) receive answer 200ok with SDP from BOB_ASIDE
c) build the INVITE with sdp from BOB and send to ALICE_BSIDE
d) ALICE will answer the INVITE with 200okSDP.
e) sending
From: ext Iñaki Baz Castillo [mailto:i...@aliax.net]
Sent: Tuesday, July 26, 2011 5:36 AM
To: Pavesi, Valdemar (NSN - US/Irving)
Cc: ext Romel Khan; sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] BYE before call answer
2011/7/25 Pavesi, Valdemar (NSN - US/Irving) :
> BYE
BYE must be used to terminated the calls after answer.
CANCEL must be used to terminated the calls before answer.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Romel Khan
Sent: Monday,
Hello,
A) SIP-I have the body SDP and plus the body ISUP. ( IAM---> trunk- TDM
---> MGC invite/sdp/iam --> MGC ---> TDM --- trunk --- IAM ( I am
here again))
INVITE/SDP/IAM
SIP-T For ISUP-SIP Interconnections or SIP-I are similar :
"
SIP-I and SIP-T refer to two very similar approach
Sdp is not part of RFC3261.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Worley, Dale R (Dale)
Sent: Thursday, June 30, 2011 9:51 AM
To: Johnson, Michael A; sip-implementors@lists.cs.columbi
...@aliax.net]
Sent: Tuesday, June 14, 2011 5:12 PM
To: Pavesi, Valdemar (NSN - US/Irving)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] release transaction after INVITE/100trying...
2011/6/14 Pavesi, Valdemar (NSN - US/Irving) :
> In case of UAS after receive INVITE and s
Hello ,
In case of UAS after receive INVITE and send 100-trying after 200 msec , if
there is no provisional response how UAS will release the transaction ?
Rosenberg, et. al. Standards Track [Page 1
Reason: X.int ;reasoncode=0x0603;add-info=0294.00AF.
Reason: Q.850 ;cause=41
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
VAN GEEL Jan (SDV/PSE)
Sent: Friday, May 20, 2011 3:57 A
, Valdemar (NSN - US/Irving)
Sent: Monday, May 02, 2011 9:07 AM
To: sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] SIP TCP UDP missing ACK_UDP
Hello,
I do have a problem with SIP TCP+UDP.
The client is able to receive INVITE by UDP or TCP.
As the INVITE is bigger then 1300 bytes
Hello,
I do have a problem with SIP TCP+UDP.
The client is able to receive INVITE by UDP or TCP.
As the INVITE is bigger then 1300 bytes the UAS is sending it by TCP.
All responseS from client will be based on via_tcp ( 100/100/200) and the
contact from 200ok_sdp don’t have the transport def
Take a look into the call flow and then you will find out when you have
to drop the dialog after send the REFER.
From: ext isshed [mailto:isshed@gmail.com]
Sent: Tuesday, April 26, 2011 12:58 PM
To: Pavesi, Valdemar (NSN - US/Irving)
Cc: Worley, Dale R (Dale); sip-implementors
Hello,
There are two type of call transfer.
A) call transfer attended
http://www.tech-invite.com/Ti-sip-service-05.html
b) call transfer unattended
http://www.tech-invite.com/Ti-sip-service-04.html
Regards!
Valdemar
-Original Message-
From: sip-implementors-boun...@lists.cs.colum
ce I have attached my cancel.txt fiel here.
Best Regards,
Ckumar.
--- On Wed, 20/4/11, Pavesi, Valdemar (NSN - US/Irving)
wrote:
From: Pavesi, Valdemar (NSN - US/Irving)
Subject: Re: [Sip-implementors] SIP request Cancel Issue
To: "ext Bre
ws.
> -Original Message-
> From: ext Brett Tate [mailto:br...@broadsoft.com]
> Sent: Wednesday, April 20, 2011 10:12 AM
> To: Pavesi, Valdemar (NSN - US/Irving);
> sip-implementors@lists.cs.columbia.edu
> Subject: RE: [Sip-implementors] SIP request Cancel Issue
>
> &g
.
-Original Message-
From: ext Brett Tate [mailto:br...@broadsoft.com]
Sent: Wednesday, April 20, 2011 10:12 AM
To: Pavesi, Valdemar (NSN - US/Irving);
sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] SIP request Cancel Issue
> This via with rport=5060 is the one generated
exa=no
a=ptime:40
a=sendrecv
a=rtpmap:101 telephone-event/8000
-Original Message-
From: ext Brett Tate [mailto:br...@broadsoft.com]
Sent: Wednesday, April 20, 2011 9:49 AM
To: Pavesi, Valdemar (NSN - US/Irving);
sip-implementors@lists.cs.columbia.edu
Subject: RE: [Sip-implementors] SIP
inal transaction cancelled and SHOULD destroy the client
transaction handling the original request.
Thanks!
Valdemar
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Pavesi, Valdema
Ckumar.,
I think the client is terminating the dialog after receives a 200OK for CANCEL.
Normally we do have:
+++
Uac > UAS
invite-->
<100/180/
---cancel--(cseq cancel)-->
<--
Hello Md Faruk Apel Chowdhury ,
+
http://tools.ietf.org/html/rfc3261#section-8.1.1.8
8.1.1.8 Contact
The Contact header field provides a SIP or SIPS URI that can be used
to contact that specific instance of the UA for subsequent requ
If you don't want rtcp . I think you have to disable it on server side.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext Siga
Sent: Monday, March 14, 2011 11:39 AM
To: Schwarz, Albrecht (Albre
Hello.
RTP Tools (Version 1.18)
The authors cannot provide support for compiling or running the
rtptools. We will gladly accept bug fixes, but all other email regarding
the rtptools will be ignored.
Description
The rtptools distribution consists of a number of small applications
that can be used
Hello,
Maybe the question could be more clear .
UAC > sending REGISTER to ---> UAS.
The response can be : 4XX, 5XX, 6XX ,2xx
++
If you send too muck REGISTER to the server , the response can a
overload like 503.
If you send a bad request , then the response will be 400-bad request.
Thanks!
-Original Message-
From: ext Iñaki Baz Castillo [mailto:i...@aliax.net]
Sent: Tuesday, February 22, 2011 8:45 AM
To: Pavesi, Valdemar (NSN - US/Irving)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] OPTIONS and 200ok with TO header without TAG.
2011
Why?
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Murad Ali
Sent: Tuesday, February 22, 2011 9:44 AM
To: Sip-implementors@lists.cs.columbia.edu
Subject: [Sip-implementors] Please stop sendi
Hello,
200ok --> To:
Do you see it as a problem ?
OPTIONS sip:atcavolt...@volte.com SIP/2.0
Via: SIP/2.0/UDP 10.48.6.189:4590;branch=z9hG4bK961069875.576.48
From: ATCAvolte08 ;tag=576.961069875.47
To:
Max-Forwards: 30
Contact:
Content-Type: application/sdp
Call-ID: 0077-0240-3948C73
Tabt.
It is simple to be done.
Dave-uac will accept request just from Dallas-uas where it is REGISTERED.
Regards!
Valdemar
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext Couret
Tabt
Se
You can use History-Info header to store your history.
INVITE sip:1B at example.com
History-Info: ;index=1,
;index=1.1,
;index=1.2
Valdemar
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implem
I think you have to go the specific TAC.
-Original Message-
From: ext Nikos Leontsinis [mailto:leontsi...@gmail.com]
Sent: Friday, February 11, 2011 5:25 PM
To: Pavesi, Valdemar (NSN - US/Irving)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] Dynamic
Hellp,
Mss ?
===
I do have it working , just as example:
Put(scrVarS_SIPRequestINVITE_SIMPLECALL);
CHANGE_LINK_IF_BIGGER(750,1);
Send;
===
Put the invite into buffer and CHANGE_LINK_IF_BIGGER if it is bigger
then 75
Hello,
Thinking about SDP. Is the session Id AND Session Version changed for
183 and 200ok ? If yes then the the sdp to be used is the latest one (
and the solution to have ringing back tone on 183 is very old).
Session ID: 0
Session Version: 0
+++
v=0
o=- 0 0
Then you just don't send the auhentication header for the subsequent
REQUESTS.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
Wyne Wolf
Sent: Tuesday, January 25, 2011 10:37 AM
To: Worley
yes , looks like you are trying to bypass the nonce and this way
bypass the authentication ?
The nonce for INVITE was created maybe like : TIME_STAMP+ BLABLA+
INVITE + RANDON + ... ( and with expiration timer like 5 minutes)
Wolf my response for your first question was just how to build the
In reallity all REQUEST will be challenged.
If BYE will be send after your nonce-expiration-timer then a new
challenge must be done.
BYE --->
< 401 Unauthorized
BYE with auth --->
< 200ok
The same for PRACK ,UPDATE ...
-Original Message-
From: sip-implementors-boun...
yes, for me the secret is equal UserMD5.
From: ext Wyne Wolf [mailto:sip@gmail.com]
Sent: Monday, January 24, 2011 5:03 PM
To: Pavesi, Valdemar (NSN - US/Irving)
Cc: sip-implementors@lists.cs.columbia.edu
Subject: Re: [Sip-implementors] What parameter
Hello,
See the MD5_A2Str where you must specify the request ( REGISER ,INVITE
,BYE)
A) CRYPT_MD5
MD5_A1Str := CRYPT_MD5 (UserMD5 & ":" & RealmStr & ":" &
UserMD5, FMT_LOWERCASE);
MD5_A2Str := CRYPT_MD5 ("REGISTER:" & UriStr,
FMT_LOWERCASE);
MD5_ResponseStr
Hello,
10.200.100.1:TCP- 15000 -- Invite with SDP --> 10.200.100.2:TCP - 5060
10.200.100.2:TCP- 5060 -- 200 OK --> 10.200.100.2:TCP - 15000
10.200.100.1:TCP- 16000 -- Invite with SDP --> 10.200.100.2:TCP - 5060
10.200.100.2:TCP- 5060 -- 100 Trying --> 10.200.100.2:TCP - 15000
Your Session Borde
please see the example:
__
UNREGISTER:
REGISTER sip:volte.com;transport=udp SIP/2.0
Via: SIP/2.0/UDP 10.48.6.189:3542;branch=z9hG4bK262236593.188.0
From: ATCAvolte08 ;tag=188.262236593.0
To: ATCAvolte08
Contact:
;+g.3gpp.smsip;+g.oma.sip-im;+g.3gpp.c
s-voice
Hello,
A lot of possibilities. But first try on wireshark decode as RTP . (
mark the udp packet , mouse right click and decode as RTP).
Best Regards,
Valdemar Pavesi
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.c
Hello,
The SIP recommendation tell us that the port must be present if it is not a
default port (5060).
INVITE sip:atcavolt...@10.48.6.189:1872;tgrp=CGR01REM;trunk-context=volte.com
SIP/2.0
To:
From:ATCAvolte08 ;tag=3012.25153171
We can sell to you ... We do have a solution for 500k sip subscribers
(udp/tcp/tls) and you can have 6000 locations with centrex and you can
have
Any number plan specific to each location. Then you can dial 3,4 ,6 ,10
, digits
-Original Message-
From: sip-implementors-boun...@
Hello,
RTP Dynamic Payload and Static Payload
Static payload is payload which is defined in the IANA rtp parameter document
An example of a static payload type is a-law PCM coded single channel audio
sampled at 8KHz. This is completely defined in the RTP Audio/Video profile as
payload type 8
Hello,
Maybe you are talking about the ReleaseCause that you must write on CDRs
When there is a SIP response 200ok +bye . The ReleaseCause must be 16 (
normal call clearing)
If the server receives or send 4xx,5xx,6xx then the reason could be
31,40,41... You have to look at
http://www.cs.colum
Client Errors ---> 4xxx
Server Errors ---> 5xxx
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
kaiduan xie
Sent: Thursday, October 28, 2010 3:19 PM
To: praveena ss
Cc: sip-implementors@l
Sure the REFER ( from subB) will initiate a new call ( INVITE ) to subC.
-Original Message-
From: ext Nauman Sulaiman [mailto:nauman762-h...@yahoo.co.uk]
Sent: Monday, October 25, 2010 11:33 AM
To: ext M. Ranganathan; Pavesi, Valdemar (NSN - US/Irving)
Cc: sip-implementors
See here:
04. Transfer - Unattended
http://www.tech-invite.com/Ti-sip-service-4.html
And
05. Transfer - Attended
http://www.tech-invite.com/Ti-sip-service-5.html
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.colum
Or MAYBE in JAVA,C++, PERL, PASCAL
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext
WARAD, MANJUNATH (MANJUNATH)
Sent: Tuesday, October 19, 2010 7:32 AM
To: Mike Coffee; Fruchtalle Siga
C
Yes you can send by TCP/UDP/SCTP and the VIA will inform the b-side to
send the response by TCP.
Best Regards,
Valdemar Pavesi
6000 Connection Drive
Irving, TX 75039, USA
Mob: +1 561 699 5166
valdemar.pav...@nsn.com
http://www.nokiasiemensnetworks.com/
-Original Message-
From: sip
To hold the call you must send a re-INVITE to the same DIALOG ( it means
same callid , to-tag and from-tag)
And you must inform b-side that sdp has changed ( incrementing o=- 7369
7369 ) and you must send a=inactive
INVITE
sip:hexxvolt...@10.48.6.189:3414;tgrp=CGR01REM;trunk-context=volte.com
Do you want put on hold/resume ?
Or
Do you want send a negative test without sdp-m-parameter. ?
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of ext Tom
Kristensen
Sent: Friday, September 17, 2
If the request is a retry of a previous request that was responded to
with a 4xx status code, it SHOULD be higher than the number used in the
previous attempt.
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-implementors-boun...@lists.cs.columbia.edu]
200ok with Expire time=0 for REGISTER just means that there is no
resource created on Server and then the client is not registered.
Normally this is the way that we have to delete the old registration
.(REGISTER with expire =0).
200ok with expire time=0 for SUBSCRIBE means that there is no dialo
The expires header is used to refresh a created dialog ( sending another
INVITE or UPDATE)
In this case INVITE/100 and UAC dies, all provisioning response are not
mandatory , then you can clean the context after the answer-timeout ( 90
seconds).
Regards!
Valdemar
-Original Message---
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