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Klaus Darilion | 1 Dec 2008 09:59
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Re: SIP Security Guidelines

Hi Dan!

How do you want to create the this document? Privatly in a closed group 
or public (e.g. like SIPconnect)?

regards
klaus

Dan York schrieb:
> Klaus,
> 
> Those are the main public documents I am aware of. I am the chair of the 
> VoIP Security Alliance (VOIPSA) Best Practices project which is charged 
> with developing these sort of guidelines but after an initial start a 
> while back, the project hasn't progressed. I am looking to re-start it 
> in early 2009, but until that document emerges the ones you mention are 
> the ones I refer people to. Various vendors have their own documents, 
> naturally.
> 
> Dan
> 
> -- 
> Dan York
> 
> On Nov 28, 2008, at 9:39 AM, Klaus Darilion 
> <klaus.mailinglists <at> pernau.at> wrote:
> 
>> Hi!
>>
>> I am trying to find publications about SIP/VoIP security (risks,
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Tamas Kocsis | 1 Dec 2008 12:05
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Develop own SIP application that runs IP Phone

Hi!

Is it possible to change the default application (sip.ld on Polycom
phones) which works the phone to a newly developped one (developped with
an appropriate tool) ?
We use Polycom SP Phones and my betters want to known if it is possible.
Or if it not - because probably i wont get the source code from the
vendor -, is there any open source like IP phone, which can be
customized/reprogrammed like this ? Do you know about any applicable
solution ?

Thanks,

Tamas
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Victor Pascual Ávila | 1 Dec 2008 12:09
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Re: SIP Security Guidelines

Hi Klaus,

On Fri, Nov 28, 2008 at 3:39 PM, Klaus Darilion
<klaus.mailinglists <at> pernau.at> wrote:
>  and some thesis about VoIP security

Any recommended reference?

Cheers,
--

-- 
Victor Pascual Ávila

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Padmaja | 1 Dec 2008 13:18
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Local ringback vs Early media ring back

Hi all,

This is regarding the local ringback tone generation vs playing early media
as per RFC 3960. I am, not clear on a few points-

1. When should the Source UA start playing the local ring back or early
media? I understand that the early media can start flowing as early as the
dest receives the invite. So if RTP packets are received by the source
before any 1xx response, can this be treated as early media and the source
UA could start playing it or should it wait for the RTP packets after the
18x response?

2. What is the bearing of the SDP received/absent in the 18x response on
choosing between local ring back and early media play back in the source UA?

3. If there is no 18x response from the destination but a direct 200 Ok
after a delay, what should be played by the source UA?

4. Can the source UA switch between initially playing local ring back and
then switching over to early media/ announcements and vice versa?

Thanks in advance,
Padmaja
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Iñaki Baz Castillo | 1 Dec 2008 14:41
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Re: Software to create SIP flows?

El Domingo, 30 de Noviembre de 2008, Anders Kristensen escribió:
> http://sourceforge.net/projects/callplot

Thanks, I'll try it :)

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Iñaki Baz Castillo

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Robert Sparks | 1 Dec 2008 17:01
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Re: Why so many implementations expect "; lr=on" or "; lr=yes" instead of just "; lr"?

You were correct to suspect a historic reason.

There was an implementation many years ago from a larger company that  
had this "lr=on" problem.
If memory serves, it would crash if you sent it a URI with the lr  
parameter and no value, and it took
several months for the bug to get fixed. At the time, lots of people  
wanted to be sure they worked
with that implementation, so a lot of code that added "=on" or "=true"  
got spread around.
As far as I know, that bug is long long gone, but the field full of  
workaround code hasn't been cleaned up.

RjS

On Nov 26, 2008, at 3:42 PM, Iñaki Baz Castillo wrote:

> Hi, AFAIK loose routing is exclusively defined in RFC 3261, and it's  
> clear and
> explained in *all* the examples and BNF section that "lr" has no  
> value:
>
>  lr-param          =  "lr"
>  <sip:proxy1>,<sip:proxy2>,<sip:proxy3;lr>,<sip:proxy4>
>
> So I wonder why there are various SIP devices that only do loose  
> route if the
> Route/Record-Route header has a parameter like:
>  ;lr=on
>  ;lr=yes
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Dan York | 1 Dec 2008 21:40
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Re: SIP Security Guidelines

Klaus,

On Dec 1, 2008, at 3:59 AM, Klaus Darilion wrote:

> Hi Dan!
>
> How do you want to create the this document? Privatly in a closed  
> group or public (e.g. like SIPconnect)?

Oh, it's fully public - there's a mailing list you can join here:

http://www.voipsa.org/Activities/bestpractices.php

It's just been dormant for a while due to a combination of factors.

My goal is to get it going again in early 2009 and see if we can't get  
the document created quickly.

You, and anyone else on the list, are welcome to join.

Regards,
Dan

--

-- 
Dan York, CISSP, Director of Emerging Communication Technology
Office of the CTO    Voxeo Corporation     dyork <at> voxeo.com
Phone: +1-407-455-5859  Skype: danyork  http://www.voxeo.com
Blogs: http://blogs.voxeo.com  http://www.disruptivetelephony.com

Build voice applications based on open standards.
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cool goose | 2 Dec 2008 07:32
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SIP stack

Hi Everyone,

I am new to the SIP and just started reading RFC 3261. I  am planning to
write my own SIP stack with minimum functionality (setting up a basic call
and tear down). Can someone point me to the right resources that can assist
me in this? Also, can someone explain what I needs to be considered in the
component architecture for SIP stack?

Thank You,
CoolGoose.
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Somesh S. Shanbhag | 2 Dec 2008 07:46
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Re: SIP stack

There are some SIP stacks available open source. You can take them as reference.

reSIProcate
VOVIDA
oSIP

- Somesh

-----Original Message-----
From: sip-implementors-bounces <at> lists.cs.columbia.edu on behalf of cool goose
Sent: Tue 12/2/2008 12:02 PM
To: sip-implementors <at> lists.cs.columbia.edu
Subject: [Sip-implementors] SIP stack

Hi Everyone,

I am new to the SIP and just started reading RFC 3261. I  am planning to
write my own SIP stack with minimum functionality (setting up a basic call
and tear down). Can someone point me to the right resources that can assist
me in this? Also, can someone explain what I needs to be considered in the
component architecture for SIP stack?

Thank You,
CoolGoose.
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Alex Balashov | 2 Dec 2008 08:37
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Re: SIP stack

cool goose wrote:

> I am new to the SIP and just started reading RFC 3261. I  am planning to
> write my own SIP stack with minimum functionality (setting up a basic call
> and tear down). Can someone point me to the right resources that can assist
> me in this? Also, can someone explain what I needs to be considered in the
> component architecture for SIP stack?

Best of luck, but do not confuse this for a trivial feat.

--

-- 
Alex Balashov
Evariste Systems
Web    : http://www.evaristesys.com/
Tel    : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (706) 338-8599
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