From: sip-implementors-boun...@lists.cs.columbia.edu
[sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Alex Bakker
[intrepid2...@gmail.com]
To:
___
To summarize and extend what others have said:
T
Anjana Arora wrote:
> Hi,
>
> As per RFC 3262, Sec 5 "The Offer/Answer Model and PRACK"
>
> “If the INVITE contained an offer, the UAS MAY generate an answer in a
> reliable provisional response (assuming these are supported by the UAC). That
> results in the establishment of the session bef
I agree with Tomasz - the 180 is the first reliable response, and so
must contain an offer.
Thanks,
Paul
Tomasz Zieleniewski wrote:
> Hi Carl,
>
> In Your case this flow is wrong, if there is no SDP in the initial INVITE,
> UAS must
> send and offer in the first reliable respons
2010/5/19 RAVI KUMAR :
> Now my question is what should be ideal behaviour in this case ?
>
> 1. we should not expect two different call-ID as a part of same invite
> message at the first point itself as callID is unique for the call? We can
> say invite itself is not proper ?
This is invalid. 40
Hi All,
I am in dilemma what should be ideal behaviour in this case.
I am getting two different call-id in single invite message . Lets say two
callid call-ID A and call-ID B as a part of same invite message.
Currently gateway while sending response it is concatenating both callid
(Call-ID A,B).
Thanks Dale for the reference. I will refer RFC 3263 for my query and come
back in case of any issue.
Best Regards,
Vivek Batra
-Original Message-
From: WORLEY, Dale R (Dale) [mailto:dwor...@avaya.com]
Sent: Wednesday, May 19, 2010 12:57 AM
To: Vivek Batra; sip-implementors@lists.cs.col
Thank you for this link!
Cheers,
Alex
2010/5/19 Singh, Indresh (NSN - US/Boca Raton)
> Ideally the user=phone should be part of the SIP URL. So it should be
> within the bracket of from and to header
>
> To:
>
> Look in the grammar and syntax section of rfc3261
>
> ---
2010/5/19 Victor Pascual Avila :
> sip:ip1 -> Send to ip1, port 5060 over UDP
>
> sip:ip1:portA -> Send to ip1, port A over UDP
>
> sip:ip1;transport=x -> Send to ip1, port 5060 over transport x
>
> sip:ip1:portA;transport=x -> Send to ip1, port A over transport x
>
> sip:FQDN -> Resolve by D
BTW, that particular URI is not completely correct because the 'user=phone'
indicates that the userinfo in the URI is a phone number, but there is no
userinfo part in that URI.
cheers,
(-:bob
-Original Message-
From: Bob Penfield
Sent: Wednesday, May 19, 2010 10:23 AM
To: 'Alex Bak
2010/5/19 Bob Penfield :
> Parameters inside the <> are URI parameters. Parameters outside the <> are
> Header parameters. 'user' is a URI parameter, so user=phone belongs inside
> the <>. So "To: " is correct.
Also take into account that, due to the complex SIP BNF grammar, the
following ;user=
Ideally the user=phone should be part of the SIP URL. So it should be
within the bracket of from and to header
To:
Look in the grammar and syntax section of rfc3261
---
uri-parameters= *( ";" uri-parameter)
u
Parameters inside the <> are URI parameters. Parameters outside the <> are
Header parameters. 'user' is a URI parameter, so user=phone belongs inside the
<>. So "To: " is correct.
cheers,
(-:bob
-Original Message-
From: sip-implementors-boun...@lists.cs.columbia.edu
[mailto:sip-im
Alex Bakker wrote:
> Hello,
>
> When I make a call from a Siemens handset to SJPhone (softphone), I notice
> something strange in the "To:" field in the INVITE request.
>
> This field looks like this:
>
> To:
>
> Is this really a valid notation?? I am writing a parser that reads these
>
Hello A
Hello,
When I make a call from a Siemens handset to SJPhone (softphone), I notice
something strange in the "To:" field in the INVITE request.
This field looks like this:
To:
Is this really a valid notation?? I am writing a parser that reads these
headers but I am unsure how to interpret this..
Let say transaction DO take this responsibility
When will transaction stop the retransmission of this response??
***
This e-mail and attachments contain confidential information from HUAWEI,
which is intended
Why retransmission of 200 OK response is handled by Transaction user layer ?
why not transaction layer ?
If, while in the "Proceeding" state, the TU passes a 2xx response to
the server transaction, the server transaction MUST pass this
response to the transport layer for transmi
Hi,
On Tue, May 18, 2010 at 8:15 AM, Vivek Batra
wrote:
> Is there any harm if we don't include the SIP port in Request-URI, To and
> >From field (of REGISTER etc) when SIP domain port is resolved using DNS SRV
> query?
sip:ip1 -> Send to ip1, port 5060 over UDP
sip:ip1:portA -> Send to ip
When are MGCP messages required?
In order to create RTP streams, is it not mandatory that CRCX happens once
SIP INVITE is initiated? How does this work?
I captured ethereal traces of a normal SIP call flow on machine A. I could
see RTP messages but not MGCP messages. Is it fine?
The Connection
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