Hi!
IIRC RFC3261 defines that dialog-creating requests may change the remote
target URI, e.g. reINVITE/200 may provide a different Contact URI which
updates the remote target URI.
What about SUBSCRIBE/NOTIFY? I didn't found a definition in RFC3265, but
I would suspect, as SUBSCRIBE is also a d
Hi!
How should a presence server react when it receives a tuple with an
already existing tuple-ID?
Is there any binding between the tuple-ID and the corresponding PUBLISH
SIP-ETag?
thanks
Klaus
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... correcting myself ...
Am 25.05.2010 11:40, schrieb Klaus Darilion:
> Hi!
>
> I have a scenario where a presence agents only performs registration
> (otherwise the SBC does not allow messages from this client) and sends
> PUBLISH requests.
>
> Thus, this SIP client on
Hi!
I try to map calendar entries to RPID elements - unfortunately this is
not trivial.
Calender entries can be roughly characterized with:
Subject: What is happening
Location: Were is it happening
When: Start-stop
Particpation-Status: e.g. free/tentative/busy/out-of-office
I suggest the follow
Hi!
Is there a something like a simple-implementors mailing lists which is
more appropriate for SIMPLE related question than this mailing list?
Thanks
Klaus
PS: Meanwhile I will continue in this mailing list
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Again I have to correct myself:
Am 25.05.2010 17:28, schrieb Klaus Darilion:
> I think my RPID document is malformed, as must be sub-element of
> , not of. Thus, I can not have a single person with
> 2 activities with different note.
It is allowed: according to the schema is allow
WORLEY, Dale R (Dale) wrote:
> From:
> sip-implementors-boun...@lists.cs.columbia.edu
> [sip-implementors-boun...@lists.cs.columbia.edu] On Behalf Of Klaus
> Darilion [klaus.mailingli...@pernau.at]
>
> Thus, this SIP client only sends RE
Iñaki Baz Castillo wrote:
> 2010/5/25 Klaus Darilion :
>> What about using "classes". E.g. by specifying class=calendar for person
>> elements created by the calender presence user agent the watcher might be
>> smart enough to render the data as "this is what hi
Iñaki Baz Castillo wrote:
> 2010/5/25 Klaus Darilion :
>> I think my RPID document is malformed, as must be sub-element of
>> , not of . Thus, I can not have a single person with
>> 2 activities with different note.
>
> Sorry, I didn't realize of that. Yes, AFAI
Am 25.05.2010 14:13, schrieb Klaus Darilion:
> Hi!
>
> I wonder if a RPID document without a tuple element is allowed and makes
> sense.
>
> Scenario: A presence user agent publishes a person's activities, e.g.
> retrieved from a calender. If I understand it right,
Hi!
I wonder if a RPID document without a tuple element is allowed and makes
sense.
Scenario: A presence user agent publishes a person's activities, e.g.
retrieved from a calender. If I understand it right, the tuple
represents the service. Thus, as this presence user agent does not offer
any
Am 20.05.2010 15:51, schrieb Olle E. Johansson:
> Here at SIPit we had a test involving B2BUA's in spirals and loops and other
> dangerous situations.
>
> Robert Sparks and I ended up with a discussion about how B2BUA's should
> handle Max-Forwards.
>
> http://tools.ietf.org/html/draft-marjou-s
Am 24.05.2010 15:03, schrieb Iñaki Baz Castillo:
> 2010/5/24 Olle E. Johansson:
>> I am sure that it's the only solution moving forward. And if guys like you
>> and I do not take it seriously, we'll be in serious trouble or just locked
>> in behind NATs in the service providers networks. I don'
Hi!
I have a scenario where a presence agents only performs registration
(otherwise the SBC does not allow messages from this client) and sends
PUBLISH requests.
Thus, this SIP client only sends REGISTER and PUBLISH, and is not
willing to accept any SIP requests.
Thus, I wonder if it is corre
Hi!
Consider the simple scenario: caller is dual-stacked and supports IPv4
and IPv6. When creating the SDP, the caller does not know if callee use
IPv4-only, IPv6-only or is dual-stacked too.
SIP probably is not an issue and will be done by the proxies - but what
about finding the proper IP ve
Hi!
I want to perform some voice quality tests (e.g. to evaluate how quality
decreases with load).
Are there some standards/benchmarks/best-practices fur such tests (to
get comparable results)?
I know SIPstone and SPEC-SIP - but these are purely SIP. Thus, any
pointer to such standardized tes
Answering myself:
ReTurn states to be RFC 5389 compliant.
http://www.resiprocate.org/ReTurn_Overview
regards
klaus
Klaus Darilion schrieb:
> Hi!
>
> Are there any open source STUN server which support RFC 5389?
>
> I only know stund (sourceforge) and mystun (berlios
Hi!
Are there any open source STUN server which support RFC 5389?
I only know stund (sourceforge) and mystun (berlios), but they are
rather old.
thanks
klaus
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Thanks a lot, now I understand.
regards
klaus
Paul Kyzivat wrote:
> inline
>
> Klaus Darilion wrote:
>> Hi all!
>>
>> RFC 3261 explicitly allows the having multiple contacts in a single
>> header.
>>
>> > 7.3: ... Specifically, any SIP header w
Hi all!
RFC 3261 explicitly allows the having multiple contacts in a single header.
> 7.3: ... Specifically, any SIP header whose grammar is of the form
>
> header = "header-name" HCOLON header-value *(COMMA header-value)
>
> allows for combining header fields of the same name into a com
Hi!
I try to summarize the service which can be set up by SIP, and the used
protocols. Currently my list is:
Service Negotiation Protocoll
-
Audio SDP
to-tag parameter
> is compared to the local tag, and the from-tag parameter is compared
> to the remote tag.
to-tag ic compared to local tag (at the UAS), thus from my understanding
to-tag should be a, and from-tag should be b.
regards
klaus
>
> thanks
> Abhishek
>
&
Hi!
I try to understand how to format the Refer-To header. Consider the
following scenario: (ltag = local tag, rtag = remote tag)
A B D
ltag: a b c d
rtag: b a d c
Now, B sends REFER to D:
REFER sip:D
Iñaki Baz Castillo wrote:
> 2009/2/26 Olle E. Johansson :
>> This is a problem I realize at every SIPit. The implementations are far away
>> from the IETF world. And the gap doesn't seem to close.
>>
>> Basic stuff like DNS is not understood or used by many SIPit attendees so
>> even trying to ment
I will try to summarize:
If Alice offers codecs X and Y, and Bob answers with X and Y, both are
allowed to send with any of these codecs and must be prepared to receive
any of these codecs. Thus, asymmetric codecs may happen.
regards
klaus
Maxim Sobolev wrote:
> Sorry, you guys were right, I w
Iñaki Baz Castillo schrieb:
> 2009/2/25 Klaus Darilion :
>> I think if Alice announces G711 and G729, and Bob answers with G.729,
>> Alice must send with G729 and Bob can send with G711 too. Is this correct?
>
> If Bob answers G729 then Alice expects that Bob will send G729
Stephane van Hardeveld schrieb:
>
> - Original Message - From: "Klaus Darilion"
>
> To: "sip-implementors"
> Sent: Wednesday, February 25, 2009 9:29 AM
> Subject: [Sip-implementors] asymmetric audio codecs
>
>
>> Hi!
>>
>&g
Maxim Sobolev schrieb:
> Klaus Darilion wrote:
>> Hi!
>>
>> For a certain application where uplink is low bandwidth and downlink
>> is high bandwidth I want to use the best available codec - ie. up
>> G729, down G.711.
>>
>> How can I setup such an
Hi!
For a certain application where uplink is low bandwidth and downlink is
high bandwidth I want to use the best available codec - ie. up G729,
down G.711.
How can I setup such an asymmetric session?
eg.
high down
Alice Bob
up low
I think if Alice announces G711
Scott Lawrence schrieb:
> On Thu, 2009-01-29 at 15:36 +0100, Johansson Olle E wrote:
>>> Don't use registration.
>>>
>>> Provision the DIDs, then all normal targeting mechanisms work just
>>> fine.
>>>
>>> Someone will say "but what about dynamic addresses" - don't use a
>>> dynamic address for
Johansson Olle E schrieb:
>
> 29 jan 2009 kl. 10.15 skrev Theo Zourzouvillys:
>
>> On Thu, Jan 29, 2009 at 8:53 AM, Klaus Darilion
>> wrote:
>>
>>> Problem: How to signal the called phone number? Usually the called phone
>>> number is in the RURI
Iñaki Baz Castillo schrieb:
> El Jueves, 29 de Enero de 2009, Klaus Darilion escribió:
>
> a) The PBX could register as many times as DID has assigned, using the
> appropiate DID for each registration.
Impossible. Consider a country with an open dialing plan. e.g my company
ha
Hi all!
We recently discussed the following problem on the asterisk-dev list and
are hoping for inputs how to solve it in a standard conform way.
Scenario: There is an enterprise using an IP PBX connected via a SIP
trunk to the service provider which handles the PSTN connectivity.
The PBX regi
Hi!
I have some issues regarding SIP in IMS I would like to discuss. Is
there somewhere an IMS dedicated SIP mailing list? Or is it OK to post
to the sip-implementors list?
thanks
klaus
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Victor Pascual Ávila schrieb:
> Hi Klaus,
>
> On Fri, Nov 28, 2008 at 3:39 PM, Klaus Darilion
> <[EMAIL PROTECTED]> wrote:
>> and some thesis about VoIP security
>
> Any recommended reference?
no
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arious vendors have their own documents,
> naturally.
>
> Dan
>
> --
> Dan York
>
> On Nov 28, 2008, at 9:39 AM, Klaus Darilion
> <[EMAIL PROTECTED]> wrote:
>
>> Hi!
>>
>> I am trying to find publications about SIP/VoIP securit
Hi!
I am trying to find publications about SIP/VoIP security (risks,
guidelines for software implementors, guidelines for service providers ...)
Currently I found:
NIST SP800-58
draft-ietf-speermint-voipthreats-00
and some thesis about VoIP security
Are there other documents regarding SIP
Vikram Chhibber schrieb:
> IMO "pres" uri is meant for presentity/watchers participants for
> "presence" and related event packages.
> "sip" scheme is more generic and may include "sip protocol"
> participants including call and "presence".
> All these schemes are meant for IP domain. TEL scheme
Paul Kyzivat schrieb:
>
> Iñaki Baz Castillo wrote:
>> 2008/10/22 Iñaki Baz Castillo <[EMAIL PROTECTED]>:
>>> Hi, SIP Presence Event Package RFC( 3856 ) says that From and To of a
>>> SUBSCRIBE cannot be a TEL URI:
>>>
>>> Section 5.
>>>
>>> SUBSCRIBE messages also contain logical identifie
Scott Lawrence schrieb:
> Whether or not to forward a SIP request that is not for your domain
> depends on your deployment model and whether or not you want to support
> end-to-end SIP calling. If you want to be able to support end-to-end
> SIP, that implies that your users can use real SIP URLs
[EMAIL PROTECTED] schrieb:
>From: Klaus Darilion <[EMAIL PROTECTED]>
>
>What is the response code for a SIP request which was routed to the
>wrong proxy, e.g. "INVITE sip:[EMAIL PROTECTED]" was wrongly routed to
>example.com SIP proxy. How should
Scott Lawrence schrieb:
> On Mon, 2008-09-22 at 16:53 +0200, Klaus Darilion wrote:
>> Hi!
>>
>> What is the response code for a SIP request which was routed to the
>> wrong proxy, e.g. "INVITE sip:[EMAIL PROTECTED]" was wrongly routed to
>> example.c
Hi!
What is the response code for a SIP request which was routed to the
wrong proxy, e.g. "INVITE sip:[EMAIL PROTECTED]" was wrongly routed to
example.com SIP proxy. How should example.com reject the request?
404 or is there a better code - something like "I am not authoritative
for this domain"?
Hi all!
I wonder about the case sensitiveness of the dialog state. What is correct:
early
or
Early
^
RFC 4235 says: '...The "state" element indicates the state of the
dialog. Its value is an enumerated type describing one of the states in
the FSM above...' T
Bob Penfield wrote:
> The INVITE transaction must still complete independent of the CANCEL.
> So the proxy would continue to re-transmit the INVITE. If the proxy
> does receive a provisional response, it would then stop
> retransmissions and send the CANCEL down stream. If timer B fired, it
> would
n cancelled and SHOULD destroy the client
> transaction handling the original request.
Ok. But that does not answer my question (as far as I can see). Should
the proxy stop retransmissions or not?
regards
klaus
>
>
> ------
> From
Hi!
Could someone help me please with this question.
Scenario: A transaction-stateful proxy forwards the INVITE request. The
proxy does not receive a provisional response, thus starts
retransmissions. Now, the caller CANCELs the call. How is the proxy
supposed to handle this? Does the proxy st
Hi!
Testing XCAP with eyebeam I see it uses the URL
http://1.2.3.4/resource-lists/users/username/resource-list.xml
to store the contact list.
I wonder where the syntax of the XCAP URL is defined - any pointers are
appreciated.
thanks
klaus
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Hi!
I try to understand if a SIP UA is allowed to send out a SSLv2
compatible ClientHello (which according to openssl man pages allows
SSLv2, SSLv3 and TLSv1).
Is a SIP UA allowed to use SSLv2 CllientHello (indicating TLSv1 support
as well) or must a SIP UA send out a TLSv1 ClientHello?
thank
Hi!
I have a question: A client (strict router) has an open
Client A -> Proxy --> client B
A calls B, B is strict router, Proxy makes Record-Routing.
If B sends a reINVITE to A, it looks like:
INVITE sip:proxy-ip;lr
Route: sip:[EMAIL PROTECTED]
If this requests gets challenged by the
Manivannan S, TLS-Chennai wrote:
> Hi everybody,
>
>
>
> Normally SIPS request will be send in TLS connection. Is it possible to
> send a SIP request in TLS connection?
Yes.
regards
klaus
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