On Thu, Jun 22, 2006 at 10:31:51AM +0530, Nataraju A B wrote:
...
> [ABN] it is very unlikely that both the proxies running the same
> software on diff machines/hardware would generate the same random
> numbers/time based random numbers since network delay could be different
> for same messages r
On Thu, Jun 22, 2006 at 10:31:51AM +0530, Nataraju A B wrote:
...
> [ABN] it is very unlikely that both the proxies running the same
> software on diff machines/hardware would generate the same random
> numbers/time based random numbers since network delay could be different
> for same messages rea
> Hie all,
> This is the first time I send a message to this mailing list, it will
be
> a general purpose question about the SIP proxy non deterministic
> behaviour;
> As far as I know, the same SIP signaling messages sent to two
different
> proxies involve different application layer behaviour, th
From: "Charles Abondo" <[EMAIL PROTECTED]>
May I have received as a UAS (last hop on a path) a Request with several
Authorisation fields (there is no Sip capable entity after the UAS? If yes,
how do I select one (realm )?
The consensus is that your UAS must (in practice) be capab
Stumer, Peggy (Com US) wrote:
> Is there any way to communicate this in SIP in a standardized way? If
> not, should there be, or, crap shoot?
Since Alert-Info is usually carried in the first INVITE message and
expected to be acted on immediately by the recipient, there is no
obvious and simple
Hello everybody! I'm new to this list and joined because I have taken a lot
of interest in SIP implementations and other VoIP-related ventures. A while
back I remember seeing threads regarding the possible implementation of an
RFC 3261 SIP stack in .NET, and I had need of one myself so I figured I'
Is there any way to communicate this in SIP in a standardized way? If
not, should there be, or, crap shoot?
-Original Message-
From: Paul Kyzivat [mailto:[EMAIL PROTECTED]
Sent: Wednesday, June 21, 2006 2:53 PM
To: Steve Langstaff
Cc: Stumer, Peggy (Com US); sip-implementors@cs.columbia.e
Steve Langstaff wrote:
> I've seen alert-info with 'info=alert-visual' used for silent ringing, but
> don't know how standard it is.
AFAIK, *nothing* about the use of alert-info is standard. Honoring it is
optional. What (if any) forms of parameters might be honored is a crap
shoot.
IMO ther
I work in a laboratory full-time testing sip implementations from various
vendors. The deal is, if they want to do business with my customer, they have
to come through this lab for a "stamp of approval" before my customer will
consider buying from them. The lab is a Telecommunications lab, leg
Hi,
I am looking for a small example code which can clearly show me
how and in what order the
basic and necassary functions call are made for a sip application,
I know about osip based partysip, vovida based apps & asterisk...But they
are full fledged applications
I wanted to learn basic
Hello all,
I have a working software with ICE support
(draft-ietf-mmusic-ice-08.txt). I would like to correspond with people
who have a working software with ICE support to launch compatibility tests.
Regards,
Guilherme Balena Versiani
ComunIP S/A
begin:vcard
fn:Guilherme Balena Versiani
n:V
Hello all,
I am trying to Subscribe to a proxy to get info regarding voicemail message
waiting info.
Here is a trace of subscribes I sent and I was never able to authorize them. I
checked nonce and response values and they are right.
What is causing the authorization to fail. SIP client also r
I've seen alert-info with 'info=alert-visual' used for silent ringing, but
don't know how standard it is.
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Stumer,
> Peggy (Com US)
> Sent: 21 June 2006 13:56
> To: sip-implementors@cs.columbia.edu
> Subje
Hi,
I am implementing a registrar and I would like help on this topic:
May I have received as a UAS (last hop on a path) a Request with several
Authorisation fields (there is no Sip capable entity after the UAS? If yes,
how do I select one (realm )?
Regards,
Charles Ab
Dale,
> The only reliable source of information is the request-URI. And in
> order to ensure that the request-URI contains enough information, the
> UA should register URIs with different user-parts for different AORs.
>From an implementation point of view that may be cumbersome though. E.g. if
Hie all,
This is the first time I send a message to this mailing list, it will be
a general purpose question about the SIP proxy non deterministic
behaviour;
As far as I know, the same SIP signaling messages sent to two different
proxies involve different application layer behaviour, this is due to
Sean,
I am not sure about the doc. But you can use the following way:
Scenario:
A->PA->PB->B
On receiving the INVITE PB should send out one INVITE with no SDP to
MMS. MMS will return the SDP(MMS) in 200 OK. Use this SDP to send a
INVITE to B. B will send its own SDP(B) in 180 ringing.(I think y
Hi all,
Is there a standardized way to indicate to the receiving device to NOT
provide ringing? I don't want to send a wav file with silence.
Thanks
Best Regards,
Peggy
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Ramachandran Iyer wrote:
> inline,,,
>
> Paul Kyzivat <[EMAIL PROTECTED]> wrote:
>
> Ramachandran Iyer wrote:
>> inline...
>>
>> "Barman, Sibon B (Sibon)" wrote: Does anybody have any idea on how to handle
>> 180 with SDP --- will it be
>> handled the same way as 183 with SDP meaning the r
Hi,
The Call-ID, To, From alongwith the CSeq headers are used as
transaction identifiers. Now, if a request arrives with one of these
headers as malformed or missing, what is the best way to handle and
respond to the request?
Even if we are able to manage to send a 400 Bad Request response, it
wi
Look at Alert-Info.
Paul
SungWoo Lee wrote:
> Thanks for the reply, Anshuman.
>
> But, rfc3960 deals with multimedia ringback in caller side.
> What I am concerning about is how to play a designated sound
> coming from media server in the callee side when it receives INVITE?
>
> Sean
>
inline,,,
Paul Kyzivat <[EMAIL PROTECTED]> wrote:
Ramachandran Iyer wrote:
> inline...
>
> "Barman, Sibon B (Sibon)" wrote: Does anybody have any idea on how to handle
> 180 with SDP --- will it be
> handled the same way as 183 with SDP meaning the ringing would be remote
> ringing rather t
> Is there any "official" limit on the size of the Call-ID header field?
Or
> any other field?
>
[ABN] RFC does not specify any limit on the size of the header fields or
parameters. It's all driven by the implementation...
RFC specify only the syntax of the message, header, and parameters
only..
Thanks for the reply, Anshuman.
But, rfc3960 deals with multimedia ringback in caller side.
What I am concerning about is how to play a designated sound
coming from media server in the callee side when it receives INVITE?
Sean
--- Original Message ---
Sender : Anshuman Rawat<[EMAIL PROTE
Hi all,
Is there any "official" limit on the size of the Call-ID header field? Or
any other field?
Thanks,
Nir.
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