Have a look at RFC 3725... You can find an exemple where you send an INVITE without SDP.
From: Rakesh Hooda <[EMAIL PROTECTED]>
To: sip-implementors@cs.columbia.edu
Subject: Re: [Sip-implementors] Sip-implementors Digest, Vol 41, Issue 32
Date: Thu, 31 Aug 2006 11:38:19 +0530
Regarding - INVITE with SDP or without SDPWhether Invite would contain SDP or not is dependent only on the UA." DISCLAIMER: This message is proprietary to Flextronics Software Systems Limited (FSS) and is intended solely for the use of the individual to whom it is addressed. It may contain privileged or confidential information and should not be circulated or used for any purpose other than for what it is intended. If you have received this message in error, please notify the originator immediately. If you are not the intended recipient, you are notified that you are strictly prohibited from using, copying, altering, or disclosing the contents of this message. FSS accepts no responsibility for loss or damage arising from the use of the information transmitted by this email including damage from virus."
For example, as per my understanding, 3GPP phone would always send SDP
in invite, would always request precondition. But phones behind rgw/AGCF may
not send SDP in invite.
There is no rule as to when SDP should be sent in invite and when not. It is up
to the implementation of UA and moreover requirement of the network which
defines whether invite would contain SDP or not.
We should always be ready to handle both the cases for all scenarios/services.
Regards
Rakesh Hooda
"None of us is as smart as all of us"
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Date: 08/30/2006 08:53PM
Subject: Sip-implementors Digest, Vol 41, Issue 32
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Today's Topics:
1. SIP INVITE with SDP and without SDP (Sarju Garg)
2. Response to Challenge in 401 (Sreejesh)
3. Re: Authorized REGISTER Request (and 401Response) (Benny Prijono)
4. Re: Response to Challenge in 401 (Scott Lawrence)
5. Re: Maximum RTCP report interval (Mahipati Deshpande)
6. Re: Authorization header field inREGISTERrequest
(Scott Lawrence)
7. Re: SIP INVITE with SDP and without SDP (Nittin Dutt)
8. Authorized REGISTER request amd 401 Reponse(PROBLEM
SOLVED!!!) ([EMAIL PROTECTED])
9. Contact header in re-INVITE message ([EMAIL PROTECTED])
10. Re: Reg Removal of Nondefault Port and Transportbyproxy
(Bala Neelakantan)
----------------------------------------------------------------------
Message: 1
Date: Wed, 30 Aug 2006 12:14:11 +0530
From: "Sarju Garg" <[EMAIL PROTECTED]>
Subject: [Sip-implementors] SIP INVITE with SDP and without SDP
To: "Sarju Garg" <[EMAIL PROTECTED]>,
<sip-implementors@cs.columbia.edu>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; format=flowed; charset="iso-8859-1";
reply-type=response
Hi all,
Are there any guidelines when to use INVITE with SDP or without SDP? If
there we want to make a UA to UA call, then should the UAC should send
INVITE with SDP or without SDP? Is there any specific cases when to use
which implementation.
Regards
Sarju
# 9810304396
------------------------------
Message: 2
Date: Wed, 30 Aug 2006 13:36:27 +0530
From: Sreejesh <[EMAIL PROTECTED]>
Subject: [Sip-implementors] Response to Challenge in 401
To: Sip-Implementors <sip-implementors@cs.columbia.edu>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi All,
How should the values of the following header parameters of
WWW-Authenticate be updated in response to 401 challenge:
1. username
[are the following values possible : sip:[EMAIL PROTECTED],
[EMAIL PROTECTED] or simply xx ]
2. uri
[Can it be the Uri of the user trying for authentication instead
of the request-uri].
What is the importance of the above two fields in Authentication by Server.
Regards,
Sreejesh
*
*
------------------------------
Message: 3
Date: Wed, 30 Aug 2006 09:07:01 +0100
From: Benny Prijono <[EMAIL PROTECTED]>
Subject: Re: [Sip-implementors] Authorized REGISTER Request (and 401
Response)
To: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Cc: sip-implementors@cs.columbia.edu
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
I see that you've been stucked on this for a while now, so let me try
to give it a shot.
[EMAIL PROTECTED] wrote:
> I just want to know whether the sequence of sub fields like
> cnonce,uri,nonce etc in "Authorization" header matter?
No, the order of the parameter doesn't matter. But be careful of these:
- some parameter values need to be quoted and some are not (see the
ABNF grammar in RFC 2617), so make sure you follow the correct rule
because some servers are picky about this.
- the presence "qop" requires sending additional parameters (for
example "cnonce" and "nc") which normally MUST NOT be sent.
> Also I m getting"stale=false","qop=auth" etc in 401 response.
> Hence it is must to send qop in authorized REGISTER request. And I m
> calculating response using MD5 algorithm.
The "stale=false" indicates that the response digest (the "response"
parameter in your Authorization header) is incorrect. Be aware that
when "qop" parameter is present, the digest is calculated differently
than when "qop" is not present.
When "qop" is present, your response should be calculated as:
response = MD5(ha1 ":" nonce ":" nc ":" cnonce ":" qop ":" ha2)
while normally, when "qop" is not present, your response will be
calculated as:
response = MD5(ha1 ":" nonce ":" ha2)
where:
ha1 = MD5(username ":" realm ":" password)
ha2 = MD5(method ":" req_uri)
Also be careful when calculating the digest, make sure you properly
un-quote the quoted parameters before feeding it to your digest
calculation.
>
> Can anybody tell me why server is sending 401 response again instead of
> "200OK"?
There are lots of things that can go wrong, and from where I'm sitting
now it's just impossible to determine what exactly has caused the failure.
But my suggestion now is to try your authentication function with
other server implementations that don't require "qop" in the
challenge, just to see if your base authentication implementation
(without "qop") works.
cheers,
-benny
------------------------------
Message: 4
Date: Wed, 30 Aug 2006 07:20:46 -0400
From: Scott Lawrence <[EMAIL PROTECTED]>
Subject: Re: [Sip-implementors] Response to Challenge in 401
To: Sreejesh <[EMAIL PROTECTED]>
Cc: Sip-Implementors <sip-implementors@cs.columbia.edu>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain
On Wed, 2006-08-30 at 13:36 +0530, Sreejesh wrote:
> Hi All,
>
> How should the values of the following header parameters of
> WWW-Authenticate be updated in response to 401 challenge:
>
> 1. username
> [are the following values possible : sip:[EMAIL PROTECTED],
> [EMAIL PROTECTED] or simply xx ]
That's a local policy matter for the server; it can be whatever the
server wants it to be, and may or may not share any common components
with the uri.
> 2. uri
> [Can it be the Uri of the user trying for authentication instead
> of the request-uri].
It must be exactly the uri that the client sent in the request-uri.
> What is the importance of the above two fields in Authentication by Server.
The username identifies the party being authenticated, and the uri is
part of what binds the authentication to the specific message.
--
Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:[EMAIL PROTECTED]
sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX
Chief Architect - Pingtel Corp. http://www.pingtel.com/
------------------------------
Message: 5
Date: Wed, 30 Aug 2006 12:31:58 +0100 (BST)
From: Mahipati Deshpande <[EMAIL PROTECTED]>
Subject: Re: [Sip-implementors] Maximum RTCP report interval
To: sip-implementors@cs.columbia.edu
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=iso-8859-1
Hi,
AFAIK, RTCP interval should not be calculated using amount of data that is sent
on a session (that is actual bandwidth consumed by session). Instead it is
calculated using amount of bandwidth a session is expected to comsume. This
depends on codec that is used.
Thanks,
Mahipati Deshpande
--- Franz Edler <[EMAIL PROTECTED]> wrote:
> Hello,
>
> I am trying to find a "maximum value for the RTCP report interval" in
> RFC3550 and RFC3551.
>
> Background of the question:
> We have unicast traffic (VoIP) and detected that in case of a long silence
> period and "comfort noise" enabled at the User Agent there is a
> corresponding gap in RTP and also RTCP packets (SR and RR). This leads to
> the termination of the session by a session border controller which is
> monitoring the existence of media (after 30secs of unidirectional silence).
>
> What I found is:
> - There is a recommended added bandwidth for RTCP (5% of session bandwidth)
> - There is a recommended partition of this bandwidth: 1/4 for senders and
> the rest for receivers.
>
> But what is in case the sender does not send because of "comfort noise" and
> a long silence period in one direction?
> Can a "maximum RTCP report interval" be derived from the above RFCs?
>
> Regards
> Franz
>
> _______________________________________________
> Sip-implementors mailing list
> Sip-implementors@cs.columbia.edu
> https://lists.cs.columbia.edu/cucslists/listinfo/sip-implementors
>
Mahipati Deshpande
__________________________________________________________
Yahoo! India Answers: Share what you know. Learn something new
http://in.answers.yahoo.com/
------------------------------
Message: 6
Date: Wed, 30 Aug 2006 08:11:46 -0400
From: Scott Lawrence <[EMAIL PROTECTED]>
Subject: Re: [Sip-implementors] Authorization header field inREGISTER
request
To: "Abu M. Muttalib" <[EMAIL PROTECTED]>
Cc: sip-implementors@cs.columbia.edu
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain
On Mon, 2006-08-28 at 19:30 +0530, Abu M. Muttalib wrote:
> Hi List,
>
> How the CNonce field is generated in context of "Authorization header field
> in REGISTER request"? Is it a random 32 byte string? Has it anything to do
> with User Name/Authorization User Name or Password or any other value?
Read RFC 2617.
The cnonce value is chosen by the UAC. It should be chosen such that no
two messages ever get the same cnonce, and such that the server cannot
predict what value will be chosen for a given message.
--
Scott Lawrence tel:+1-781-938-5306;ext=162 or sip:[EMAIL PROTECTED]
sipXpbx project coordinator - SIPfoundry http://www.sipfoundry.org/sipX
Chief Architect - Pingtel Corp. http://www.pingtel.com/
------------------------------
Message: 7
Date: Wed, 30 Aug 2006 13:25:22 +0100
From: "Nittin Dutt" <[EMAIL PROTECTED]>
Subject: Re: [Sip-implementors] SIP INVITE with SDP and without SDP
To: "'Sarju Garg'" <[EMAIL PROTECTED]>,
<sip-implementors@cs.columbia.edu>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;charset="us-ascii"
INVITE without SDP indicates that caller is not interested in media at this
moment. Purpose of this is highlighted in 3PCC (RFC 3725). Applications such
as click to dial OR conference-operator where user initiate the session but
other party takes responsibility of initiating the Media.
In INVITE without SDP scenario Callee offer (SDP) is sent in 200OK and then
Caller respond to the offer through ACK.
-Nittin Dutt
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of Sarju Garg
Sent: 30 August 2006 07:44
To: Sarju Garg; sip-implementors@cs.columbia.edu
Subject: [Sip-implementors] SIP INVITE with SDP and without SDP
Hi all,
Are there any guidelines when to use INVITE with SDP or without SDP? If
there we want to make a UA to UA call, then should the UAC should send
INVITE with SDP or without SDP? Is there any specific cases when to use
which implementation.
Regards
Sarju
# 9810304396
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------------------------------
Message: 8
Date: Wed, 30 Aug 2006 21:12:14 +0530
From: "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Subject: [Sip-implementors] Authorized REGISTER request amd 401
Reponse(PROBLEM SOLVED!!!)
To: sip-implementors@cs.columbia.edu, [EMAIL PROTECTED]
Cc: [EMAIL PROTECTED], [EMAIL PROTECTED]
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
Hi,
I m very glad to inform you that the problem I was facing last two days
is finally solved.
The problem was in calculating "response" field in "Authorization"
header of authorized REGISTER request.
Actually the "response" field of "Authorization" header is calculated
using MD5 algorithm.
This algorithm takes "Nonce" , "Nonce-Count" , "CNonce" , "qop" ,
"Method" , "Request-URI" ,"username" etc as the input parameters and
produces 128-bit "response".
Out of which "Nonce" , "qop" etc are provided by server in 401 response.
Server provides qop as a quoted string for example qop"quth".
But in authorized REGISTER request we have to put qop as unquoted string
like qop=auth.There are other fields like nonce-count and cnonce which
we have to generate.
*THE INPUT PARAMETERS OF MD5 ALGORITHM ARE ALL UNQUOTED STRINGS. AND
HERE WHERE THE PROBLEM WAS...
*I was passing "user name" and "qop" as quoted strings to the MD5
algorithm. Hence algorithm was giving me wrong "response".
Then I was putting this wrong response in Authorized REGISTER request
because of which server was again giving me 401 response.
*But finally I passed all the input paramters to MD5 algorithm as
unquoted strings and SUCCESS!!!* :-)
Anyways , the problem was not as complex as I was expecting , But I
learned a lot from it.
Thanking you for always encouraging and giving me positive responses.
Hope to get such responses in future also. ;-)
With Regards,
---Sumit 8-)
------------------------------
Message: 9
Date: Wed, 30 Aug 2006 10:13:57 -0500
From: [EMAIL PROTECTED]
Subject: [Sip-implementors] Contact header in re-INVITE message
To: sip-implementors@cs.columbia.edu
Cc: [EMAIL PROTECTED]
Message-ID:
<[EMAIL PROTECTED]>
Content-Type: text/plain; charset="us-ascii"
rfc 3261 states the following
Section 12.2 Page 75
" Requests within a dialog MAY contain Record-Route and Contact header
fields. However, these requests do not cause the dialog's route set
to be modified, although they may modify the remote target URI."
Page 75
" A UAC SHOULD include a Contact header field in any target refresh
requests within a dialog, and unless there is a need to change it,"
Does this mean that Contact header is not mandatory/required in a
re-INVITE message
or a 2xx response to a re-INVITE message.
Table 3, Section 20, page 162 shows that the Contact is mandatory in INV
and 2xx messages.
------------------------------
Message: 10
Date: Wed, 30 Aug 2006 10:20:47 -0500
From: "Bala Neelakantan" <[EMAIL PROTECTED]>
Subject: Re: [Sip-implementors] Reg Removal of Nondefault Port and
Transportbyproxy
To: "'karthik'" <[EMAIL PROTECTED]>,
<sip-implementors@cs.columbia.edu>
Message-ID: <[EMAIL PROTECTED]>
Content-Type: text/plain;charset="us-ascii"
If you read further down, the reason is given. The
was received using the port and transport indicated (explicitly or by
default) in the Request-URI, the proxy MUST strip the maddr and any
non-default port or transport parameter and continue processing as if
those values had not been present in the request.
Rosenberg, et. al. Standards Track [Page 96]
RFC 3261 SIP: Session Initiation Protocol June 2002
A request may arrive with a maddr matching the proxy, but on a
port or transport different from that indicated in the URI. Such
a request needs to be forwarded to the proxy using the indicated
port and transport.
Also refer to the Section 18.1.1 Sending Requests and see how multicast is
handled.
Thanks,
Neel
-----Original Message-----
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED]] On Behalf Of karthik
Sent: Monday, August 28, 2006 12:49 AM
To: sip-implementors@cs.columbia.edu
Subject: [Sip-implementors] Reg Removal of Nondefault Port and Transport by
proxy
hi all,
As per RFC 3261 it is given in section 16.4 that the Proxy has to remove
the Default port and transport in the Request URI when it gets an INVITE
with those parameters.
Kindly tell me what is the significance of removing those parameters
before forwarding it to next level?
Thanks in advance,
Geetha.
--
karthik
[EMAIL PROTECTED]
--
http://www.fastmail.fm- Same, same, but different.
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