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:
On Mon, Mar 21, 2011 at 9:25 PM, Steve Underwood
<steveu@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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