[Freeswitch-users] Does Freeswitch complies with SIP Connect technical recommendation?

[Kristian Kielhofner]

FreeSWITCH certainly supports most of the features/standards required
by SIPConnect 1.1 (codec support, TLS, Caller ID, etc).  It's a matter
of configuration.  It wouldn't be that difficult for someone to create
a SIPConnect 1.1 compliant SIP profile and dialplan.

12.1 - I'm not sure but doesn't sound that difficult.
14.1 - Yes (3PCC)
14.3 - Yes.  RFC 4733 is essentially the same as RFC 2833.  The
difference is that RFC 4733 requires SRTP to be in use (supported by
FreeSWITCH).

Paying to get certified, however, is ridiculous.

I have a philosophical problem with vendor and (to a lesser extent)
third party certifications.  It's a great way to still extort money
out of people when using a standard protocol.  Avaya, for example,
requires thousands of dollars a year to maintain an Avaya DevConnect
membership so you can do SIP interop with their equipment.  Hmmm, I
wonder what revenue this "certification" is supplanting for them?
Perhaps revenue they used to generate from proprietary protocols,
handsets, and extensions?

On Tue, Mar 22, 2011 at 4:23 AM, Yehavi Bourvine
<yehavi.bourvine at gmail.com> wrote:
> I skimmed through the document and it seems that FreeSwitch can be tailored
> to comply with most of the requirements by the end user. There are 3 issues
> which I am not sure about:
>
> Section 12.1 states that call transfer should be done with INVITE/Re-INVITE
> and not by REFER. From the SIP traces I've done I see that FS uses REFER to
> transfer a call.
> Section 14.1: Does FS accepts INVITE with no SDP inside?
> Section 14.3: Does FS supports RFC-4733? This section allows also RFC-2833
> for those who do not support 4733.
>
>                                       Thanks, __Yehavi:
> 2011/3/22 Michael Collins <msc at freeswitch.org>
>>
>> On Mon, Mar 21, 2011 at 9:25 PM, Steve Underwood <steveu at coppice.org>
>> wrote:
>>>
>>> I'm not clear how you can get a PBX SIP CONNECT approved. A lot of the
>>> document comes down to how you configure and use the system. Obviously,
>>> a product could present a roadblock that prevents SIP CONNECT compliance
>>> in a working setup, but I doubt that many products would do that.
>>>
>>> Characterising the SIP forum as a who's who of retarded SIP providers
>>> and PBXes is a little unfair. Practically everyone in the VoIP business
>>> is in that list. Capability certainly doesn't look like a prerequisite,
>>> though. :-\
>>
>> Hehe, true enough. Perhaps I was a little harsh. Still, you are quite
>> right about the compliance test being incredibly subjective and capability
>> not being a prerequisite. Sonus and ShoreTel are not exactly known for their
>> SIP interop features.
>> -MC
>>
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-- 
Kristian Kielhofner