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@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@freeswitch.org>
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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