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

[Yehavi Bourvine]

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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