I will be out of the office starting 27/05/2005 and will not return until
27/08/2007.
As of the 27th May I am no longer employed at MMV,business correspondence
can be directed to the Team Leader of MMV Tech Services, Justin Bree
'[EMAIL PROTECTED]'. Personal email can be sent to
[EMAIL
Jun,
Responses inline below:
Wayne,
Thanks for your reply.
By the way, I have got one more question regarding your replay.
The scenario I have mentioned is only possible with a help of the stateful
proxy. That was what you explained, wasn't it ?
Wayne: Your questions was basically
Paul,
I have a question on the signalling below. The playout from B is
indeterminate so doesn't there need to be some mechanism for B to inform
the Proxy that the playout/digit collection(whatever) has completed so that
the Proxy can terminate this session and INVITE the actual target.
Gaurav,
There are initiatives such as Key Pad Markup Language (KPML) as in
the draft RFC draft-burger-sipping-kpml which define a standard way of
communicating key pad events in SIP bodies.
- Wayne
Gaurav asked:
*
Hi,
Apart from RFC 2833, is
Pong,
I was going to ask you why the long delay between the 183 and 200 !,
I guess you beat me to it.
The trace is helpful but doesn't explain the delay. The field Resent
Packet: False might not mean much depending on where the trace was taken
and over what transport protocol the
Geetha,
I could not find the phrase that you have qouted in either of the two
sections you have specified nor elsewhere in 3261. The two sections discuss
rejecting an offer for different reasons.
Section 13.3.1.3 discusses rejection of the offer for either of the
following
Gaurav,
A very liberal description is that PSTN to SIP gateways act on
information between the two signalling protocols to enable interworking -
call setup, maintenance and termination. It sounds you have already found
the existence of documentation/RFCs such as 3398 (ISUP to SIP
Gaurav,
There have been quite a few internet drafts on application
integration in SIP both on framework and specifics such as signalling eg.
MSCML (Media Server Control Markup Language)
http://www.ietf.org/internet-drafts/draft-vandyke-mscml-06.txt
Regards - Wayne.
Gaurav wrote:
Vishal,
If the subscriber wishes to unsubscribe (this should be equivalent to
your log out) it sends a new SUBSCRIBE request with expires=0, the notifier
will send back a NOTIFY with the Subscription-State set to terminated. This
is covered in RFC3265 section 3.1.4.3 and 3.2.2.
If
Vishal,
The From and To tags are used to identify a dialog, the UAC which
forms the request will append a tag to the From field and the UAS which
responds (like your 200 for the SUBSCRIBE below) will append a tag to the
To field (which is why you don't see it in the request - SUBSCRIBE /
Udit,
I was waiting for someone more savvy than me to answer this one but I
will just have to dissapoint. I had a read of relevant section in 3261 and
3263.
If the UAC populated the sent-by field with IP address and not a port
that this will result in the UAS processing this to
Vishal,
Your further questions or postings should be made to the SIP
Implementors group mailing address and not to individuals, I am happy to
respond and try to help.
snip
suppose i am already logged in that is through the REGISTER request
/snip
Wayne - ok so the location service knows
Christian,
I didn't see a response to this query from anyone yet so here is my 2
cents.
Think of a couple of PSTN GWs that don't dynamically register
(statically defined) one experiences communication failure (path or actual
end point failure). Proxy / UA without knowing better
Taisto,
Thanks, I understand the scenario in question now. Help someone.
I think perhaps new Proxy requests *DNS iteration*, even recursive
requests from retargetting match the same 'response context' which is
running the same stateful proxy C timer (?). This timer is reset not on
Vishal,
I suspect it will depend on what message header and what you mean by
wrong. If by wrong you mean a malformed SUBSCRIBE header needed for
processing then a 400 Bad Request response with an applicable reason phrase
should be sent. If by wrong you mean the Event package is not known
Vishal,
Sorry the previous was off the mark / unhelpful, but your response
has confused me more than clarified the query. Your original posting
referred to a SUBSCRIBE request therefore I interpreted the rest as an
issue with the formation of that SUBSCRIBE request, but the example /
Diego, Prabhan,
Just a thought - have you looked at KPML ?
(draft-ietf-sipping-kpml-07).
- Wayne
*
Hi;
A quick guest,
You can send a multipart message with 3 basic parts each one with
application/dtmf-relay.
Regards
Diego B
prabhan wrote:
Kasam,
I see what you are doing and have the following two points for
discussion.
RFC 3960 discusses use of early media and if I understand correctly
makes a point that UAs might not reliably play early media over local
ringback and for this reason it is strongly recommended
Ramesh,
My thoughts on your response / question. If the a= atribute is not
grouped to a specific (valid) media codec then it is taken as applying to
all. If the attribute is not valid for a certain media eg your example of
ptime 40 not being valid for codec 4. Maybe a few things can
Ramesh,
Shouldn't it be like other media attributes and depend on it's
grouping ?, if it is grouped with the codecs a= attributes then it applies
to that codec or it can be specified before and apply to all.
Wayne
Ramesh said:
***
Hi,
When a ptime is
Alpa,
I am not worldly enough in deployments to say what is or isn't the
top couple of deployment models but suffice to say that there are obviously
more scenarios than the two outlined below, which is not a really helpful
comment !. Along your thinking what you host or not (eg gateways)
Paul,
Let me start by saying that I agree that the simplest implementation
(preferred) would be to leave the refresher parameter not specificied in
the request, it is what is recommended in the draft. But I do not think
specifying the refresher in the request is stupid and to me it seems
Alpa,
Short answer:
You will not come close to replacing the functions that CCM provides with a
CSPS, but it is possible to replace CCM with another softswitch and retain
/ use other Cisco products in this environment.
long answer:
My exposure to Cisco Call Manager (CCM) is
David,
I have also come across the same scenario. In the instance I worked
on it was a case of double jeopardy where the 486 response code back to the
B2BUA that processed calls for the domain maps this to local treatment and
therefore answers the call - terminating it on a media server
I hope this helps. The draft recommends that refresher should send the
session refresh at half the negotiated timer value, so in your example
3600 * 0.5 = 1800, but can send the refresh when it wants as long as it
completes in time.
Regards - Wayne.
Dear all,
If an UAS having sent 200OK for
Giuseppe,
My understanding - INVITE is the method for both. The term
re-INVITE applies to INVITE request within a dialog for many purposes
including placing the other party on hold as in your query.
Regards - Wayne D.
Giuseppe said:
Chaitali,
If support for it has been indicated in the signalling from the
UAS (UPDATE listed in the Allow header), the UAC can change the SDP by
sending an UPDATE request (RFC3311), in early or comfirmed dialog - as in
your example below. Or complete the initial INVITE request and then
Chiatali,
My undertsanding of the section in 3264 you are querying is that
if you offered two m= lines in the original request you need to offer at
least that in the new request and that the ordering of them is preserved,
new media offered should be appended to the previous SDP. So if
Sumit, Eugene,
Just some additional comments on the the query from Eugene.
OPTIONS message can be somewhat helpful along the lines of what Sumit was
asking beyond capability discovery, as it is processed like an INVITE
request if the user-agent is busy then you should get a '486 Busy
John,
I may have missed the point on this one. For the SIP endpoint the
Caller-ID is in the SIP signalling that sets up the call (INVITE) and the
SIP phone generates it's own tones. But then you can have a SIP endpoint
like one of the many IAD / ATAs that are out there which would need
Subhartha, Banibrata,
The 'X' in the codec definition means it is not well known /
defined standard or similar ? so if in doubt is / would be safer to define
codec properties with the attribute lines to define all properties for
these in the SDP - especially with comments such as those
Andreas,
My thouhts are if you include the a=fmtp line in the SDP as you
have illustrated below then the receiver knows you are capable of
recieving those DTMF and Tone events - but if the Answerer does not return
the a=fmtp SDP line in the response with it's supported events then
Vini,
There is no timer (I know of)for a UA being held, there are
possibly other timers running in this scenario for session refresh or RTP
timeout etc. The intermediate softswitch is a B2BUA ?(it is sending a BYE
when the B party seems unreachable ?) - more detail of the scenario would
Balu,
When reading your posting I couldn't think of a scenario that
creates a subscription within an existing dialog (I have a long way to go
trying to learn this stuff). It occured to me that perhaps this section
has been written with the premise that the SUBSCRIBE created the dialog -
Billy,
I am unsure what a MMS message is ?. But if the intent of a user
is to establish RTP then this is expressed as an offer in SDP attached to
the SIP messaging - if the far end is capable and available then this will
be communicated back to the offerer and if not that will also be
Kamran,
I spotted the following;
The realm you are using in creating the challenge response appears
to be [EMAIL PROTECTED], but the realm is the 401 Unauthorized challenge is
asterisk.
Regards,
Wayne
Kamran Ahmad [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
28/12/2004
and therefore does not breach this stipulation. Did that make sense ?.
Regard
Wayne Davies
System Support Engineer - Broadsoft
Office: +61 3 96260468
Mobile: 0417282909
email: [EMAIL PROTECTED]
Jhon miller [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
24/12/2004 06:24 PM
To: SIP_Imp
Serge,
Queries on existing SIP usage and behaviour should be sent to the
SIP Implementors group rather than the SIP IETF WG. Please follow up
responses to this thread and direct furture queries to the Implementors
list and not the IETF - thanks.
In response to the query - It
Brett,
IMO. Section 10.2 of 3264 seems to support that kind of mechanism
of placing the SDP inactive and requiring the other party to reINVITE to
activate the media but I do not think it applies in your scenario below.
The 10.2 signalling example seems to be written for initial
Ramesh,
I believe I am yet to see a 182 message used in signalling
scenarios. I believe that the answer may lie in your question.
If the message is queued then the end point is not being alerted
eg. not 'ringing' and so it would be erroneous to send a 180 Ringing back
yes ?,
David,
Scenario 1 is common with phones able to do 3way conferencing /
mixing internally, although the signalling is normally initiated from this
mixing point i.e. it is PhoneB in your diagram below. A scenario for this
is 3 way conferencing or consultative transfer. For the C party to
Joanna,
SIP preconditions if you have seen discussion about them on
implementors or read the draft would seem the best fit here. Otherwise it
may be possible to reduce the offered codec list based on some external
logic (IPCIF, IP Calculated Impairment Factor | MOS etc). If the UA is
.
Regards,
Wayne Davies
Wainwright, John [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
19/11/2004 05:45 AM
To: 'Troy Cauble' [EMAIL PROTECTED],
[EMAIL PROTECTED]
cc:
Subject:RE: [Sip-implementors] Distinctive Call ringing.
This could be the case - I
://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Wayne Davies
System Support Engineer - Broadsoft
Office: +61 3 96260468
Mobile: 0417282909
email: [EMAIL PROTECTED]
**
- NOTICE FROM DIMENSION DATA AUSTRALIA
Suheel,
I don't know about 'typical configuration' I just kind of diagreed
with the statement that HA / redundancy is not possible. Didn't this
thread start with the concept of two b2b servers which shared dialog state
information between them ? - UAC redundancy to these servers should
)
Regards,
Wayne Davies
Right! If you b2bua is used also to terminate media (e.g PTT server) than
you are in trouble, and may need to advertise new IP/port etc via Reinvite
- not sure that that is what you desired...
-uri
-Original Message-
From: Lau Jason-A13484
Sent: Wednesday
the feature back on potentially with itself as the refresher.
(Require with timer, SessionExpires 90 with a refresher of UAS ).
Wayne Davies
John Smith [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
14/11/2004 06:06 PM
To: [EMAIL PROTECTED]
cc:
Subject
will respond with an offer in
the 200 OK and then you will need to send your response in the ACK
these need to be the same codecs and o= line as the current session or you
will end up chaning the media used - which would not be the desired affect
of just a 'Contact' refresh.
Regards,
Wayne
Brett,
Your response and the section from the draft confirm my previous
point. With session expires successfully negotiated and with the UAC as
the refresher (even if it is not) if it sends a refresh, reINVITE |
UPDATE, without the Session-Expires from it's point of view it is
Sarika,
I love this question, but I don't have an answer for you - just
some thoughts. It is legit to have a REGISTER request without a contact
field so says 3261, so I believe it. Section 10.2.3 Fetching Bindings
states:
A success response to any REGISTER request
provider etc would manually
configure [EMAIL PROTECTED], [EMAIL PROTECTED] etc.
In the configurations I have seen the GW does not register on
behalf of the phones.
Wayne Davies
sunil vatnal [EMAIL PROTECTED]
29/10/2004 05:34 PM
Please respond to sunil vatnal
Sunil,
It is somewhat typical IMO that a GW device in this scenario would
be statically assigned to a user - and the server acting as a Application
Server / Registrar will hold a this static registration that will not
expire.
As an example:
When a incoming call comes
Nenad,
Setting the Session-Expires header to 0 will turn off the timer -
the call is unaffected the requirement | expectation to send a receive
session refreshes is removed. Not including Session-Expires is the same as
setting it to 0 and therefore, as described above, it will turn the
Ashish,
My understanding is that the PRACK is generated like a provisonal
response 1xx [it is just a reliable one ;-) ] and therefore in respect to
the Request-URI it will be the one that was used to create the dialog and
not the uri as defined in the contact field.
Wayne Davies
Metha,
Yes it is possible. It would be negotiated by SIP like any other
media connection, I guess a phone call may be considered 'normal' but if
you stay on Implementors long enough you will soon discover 'normal'
covers alot of territory.
Without knowing specifics I am
Sam,
Alot of SIP is based from http, RFC3261 section 22.4 defines the
digest authentication method and refers to RFC 2069 with some rules.
Regards,
Wayne Davies
sam n [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
29/09/2004 11:07 AM
Please respond to sam n
To: [EMAIL
-sipping-3pcc spec.
Wayne Davies
sam n [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
20/09/2004 03:11 PM
Please respond to sam n
To: [EMAIL PROTECTED], [EMAIL PROTECTED]
cc:
Subject:Re: [Sip-implementors] Redirecting media streams to the phone
Hi,
If i keep
will
contain a contact address of the phone (whereas the contact, VIA paramter
in the SIP header will be the Applications address). The two addresses for
signalling and media are designed to be abstracted in this way and will
enable the functionality you are requiring.
Wayne Davies
sam n [EMAIL
for a non-INVITE request
(T1, T2) presuming the connection is over UDP.
Regards,
Wayne Davies
System Support Engineer - Broadsoft
Office: +61 3 96260468
Mobile: 0417282909
email: [EMAIL PROTECTED]
Sambit [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
14/09/2003 10:11 PM
a SDP offer to gauge the UA's
ability to support a certain codec.
Regards,
Wayne Davies
Mohammed Smadi [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
02/09/2004 04:06 AM
To: [EMAIL PROTECTED]
cc:
Subject:[Sip-implementors] newbie question
hi;
i just
to be provided by another device - from rel 10 Broadsoft introduced a
conferencing server and I am sure many other vendors have a similar box
that will do the function you require.
Regards,
Wayne Davies
sam n [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
10/09/2004 09:50 AM
Please respond
the ACK to complete the
transaction - no need to send a BYE. Again if the precondition was not
mandatory there would be no need to BYE the call on indication that it
could not be met, the offerer would communicate this but complete the
transaction(s) with a 200 OK.
Regards
Wayne Davies
System
as possible.
Regards,
Wayne Davies
Idnani Ajaykumar-AIDNANI1 [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
11/08/2004 11:33 PM
To: '[EMAIL PROTECTED]' [EMAIL PROTECTED]
cc: '[EMAIL PROTECTED]' [EMAIL PROTECTED]
Subject:RE: [Sip] Some Proceeding state questions
/8000
a=ptime:25
a=rtpmap:110 telephone-events/8000
I am not familar with the ptime attribute but hope the above makes
sense.
Regards,
Wayne Davies
Roman Koverov [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
07/08/2004 01:02 AM
To: [EMAIL PROTECTED]
cc
John,
Yes, it is my understanding the Cisco Call Manager 4 will support
SIP.
Wayne Davies
Sun Technology - John PG [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
08/08/2004 02:21 AM
Please respond to sales
To: [EMAIL PROTECTED]
cc:
Subject:[Sip
please respond to resolve his posting.
Wayne Davies
sandeep chauhan [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
19/06/2004 03:17 PM
To:[EMAIL PROTECTED]
cc:
Subject:[Sip-implementors] where to put SIP package getting run time error
Hi,
I am Sandeep . I am
Ira,
There is a draft draft-ietf-sip-session-timer-xx which proposes a way of doing this. Basically the use of this feature and the timer for the interval to perform the audit is negotiated between the UAC and UAS. The failing of the audit can be used for deleting a session as per your
Mukul,
Your description of the three cases seems correct to me.
Regards,
Wayne
Mukul Purohit [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
09/06/2004 03:20 PM
To:[EMAIL PROTECTED]
cc:
Subject:Re: [Sip-implementors] Request URI for REGISTER
Hi list,
to look into it yourself - it does not follow that you escalate your query to them.
Regards,
Wayne Davies
sandeep chauhan [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
07/06/2004 06:26 PM
To:[EMAIL PROTECTED]
cc:
Subject:[Sip-implementors] sipStack instantiation
Alex,
I will have a go at answering this but will probably also get it wrong - I think there may be some confusion with the terminology. Responses inline
Regards,
Wayne.
Alex said:
I have confused myself and am hoping someone can unconfuse me.
Take a situation like this
/-
Usama,
The ACK completes the INVITE dialog and should be sent for all of the responses below - section 17.1.1 of RFC 3261. The UAS in the Invite dialog will continue to resend the final response at T1 until T2 or it receives and ACK.
Regards,
Wayne Davies
MANSOOR Usama FTRD/DMR/LON
a unicast IP address in the contact header for further dialog within the session.
Wayne Davies
Markus Hofmann [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
19/05/2004 12:45 AM
To:[EMAIL PROTECTED]
cc:
Subject:Re: [Sip-implementors] Newbie: maddr parameter
Hi
address header fields and a maddr address in the SDP.
Regards,
Wayne Davies
Hello everybody,
I have questions about the maddr parameter. I have to write a program which should have an maddr in the Request-URI. I read the RFC 3261 but it seems that I don't have an understanding for this.
What
information is held across the two Proxies then the same address can be returned in the contact field to provide in dialog failover.
Let us know if this is not in the ball park.
Regards,
Wayne Davies
The Rev [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
14/05/2004 05:36 PM
To:[EMAIL
of security ?, 3261 states that the Require and Supported headers MUST only use standard track RFC extensions.
Regards,
Wayne Davies
Kevin Bouchard [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
28/04/2004 11:31 PM
To:Phan Quang Minh [EMAIL PROTECTED]
cc:[EMAIL PROTECTED
David,
If you haven't come across it yet you should read rfc2833 RTP payload for DTMF digits.
Regards,
Wayne Davies
email: [EMAIL PROTECTED]
David Stuart [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
23/04/2004 10:56 PM
To:[EMAIL PROTECTED]
cc:
Subject
2848 - but have got it now and will have a read. I have seen some discussion in this forum on unsubscribing and the method that seemed to have more backing was to use a Subscribe message with an expires of 0. Perhaps someone else can best answer your third question.
Regards,
Wayne Davies
email
a=ptime:20
Regards,
Wayne Davies
email: [EMAIL PROTECTED]
Paul Kyzivat [EMAIL PROTECTED]
21/04/2004 07:10 AM
To:[EMAIL PROTECTED]
cc:Ramachandran Iyer [EMAIL PROTECTED], Sushil Kumar Verma [EMAIL PROTECTED], [EMAIL PROTECTED], [EMAIL PROTECTED]
Subject:Re: [Sip
Example - IPv6 OK message.
SIP/2.0 200 OK
Via:SIP/2.0/UDP .......:5060
From:sip:[EMAIL PROTECTED]
To:sip:[EMAIL PROTECTED]
Call-ID:[EMAIL PROTECTED]
CSeq:102 REGISTER
Contact:sip:[EMAIL PROTECTED]:5060;q=0.5;expires=3599
Content-Length:0
Good luck,
Wayne Davies
email
is the default.
Thanks again, regards.
Wayne Davies
email: [EMAIL PROTECTED]
Christian Stredicke [EMAIL PROTECTED]
07/04/2004 02:50 PM
To:[EMAIL PROTECTED]
cc:[EMAIL PROTECTED]
Subject:RE: [Sip-implementors] Use of Supported header field to indicate support
, but this example is an exception to the norm.
Hope this helps.
Regards,
Wayne Davies
System Support Engineer - Broadsoft
Office: +61 3 96260468
Mobile: 0417282909
email: [EMAIL PROTECTED]
Ramy Sam [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
26/03/2004 03:10 PM
To:[EMAIL
the duration of the call. There are alot of benefits in having the end UA's perform this function.
Regards,
Wayne Davies
email: [EMAIL PROTECTED]
Sushil Kumar Verma [EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED]
19/03/2004 05:16 AM
To:[EMAIL PROTECTED], [EMAIL PROTECTED]
cc
Krishna,
I am unsure of the CPC syntax but would think that the syntax rules for section 25 RFC3261 would apply and that a SIP URI would be formatted in the former rather than later eg,
From: sip:[EMAIL PROTECTED];cpc=payphone;tag=1928301774
Regards,
Wayne Davies
email: [EMAIL PROTECTED
--
| Diversion: tel:+19195551002
| ;reason=user-busy
| ;counter=4
| Diversion: tel:+19195551001
| ;reason=unconditional
| ;counter=1
Regards,
Wayne Davies
System Support Engineer - Broadsoft
Office: +61 3
of an intelligence in the proxy or UA that could redirect the incoming call to a FAX although that would be nifty.
Regards,
Wayne Davies
System Support Engineer - Broadsoft
Office: +61 3 96260468
Mobile: 0417282909
email: [EMAIL PROTECTED]
Christian Stredicke [EMAIL PROTECTED]
Sent
message. Check which environment you are in and if any services like CFNA are provisioned for the first B party.
If you believe the above may be happening look also for the diversion field in the SIP Invite to the new B party.
Regards,
Wayne Davies
[EMAIL PROTECTED]
Sent by: [EMAIL PROTECTED
field should have the actual IP address of the endpoint - although this may be re-written by a FW ALG for NAT traversal - the 'o' origin line in the SDP should includes the actual IP address of the originator and will not be changed by FW ALG etc.
Regards,
Wayne Davies
System Support Engineer
to this world. Any and all info appreciated.
Regards,
Wayne Davies
**
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