Does Freeswitch complies with SIP Connect technical recommendation?
Hello, We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the SIP CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present. Does Freeswitch fullfills all the recomendations there? Thanks, __Yehavi:
Most likely no On Mon, Mar 21, 2011 at 11:46 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com> wrote:
Hello,
We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the SIP CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present.
Does Freeswitch fullfills all the recomendations there?
Thanks, __Yehavi: _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
No, we have not paid them thousands of dollars for them to say that we are "SIP CONNECT Compliant." That being said, I read through some of the 1.1 specs. I didn't see anything that was egregious, but I also did not look at all 45 pages. This much I can tell you: to be "SIP CONNECT Compliant" you would need to program your dialplan for some specifics. Here's an example: Section 13 of the spec says: *13 Emergency Services* The SIP-PBX MUST have a dial plan that recognizes emergency calls. This will differ from one country to the next and is a function of the administrator's dialplan configuration. There are no inherent limitations in FreeSWITCH to prevent you from being up to spec on this. However, out of the box with a default installation you won't have this already configured. My guess is that we could write tests that demonstrate that FreeSWITCH *could* be SIP CONNECT compliant, however I doubt that would be of much value because you have to cough up a minimum of $2500 to have their seal of approval. Beyond that look at their membership list: http://www.sipforum.org/component/option,com_fullmember/Itemid,195/ It's a who's who of retarded SIP providers and PBXes. They've got Sonus on their list for cying out loud! That alone should invalidate the entire SIP CONNECT venture IMHO. So, I would answer your supplier this way: As far as I know there aren't any limitations in FreeSWITCH that would prevent it from complying with the SIP CONNECT recommendations. Oh, and GLWT! :) -MC On Mon, Mar 21, 2011 at 11:46 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com
wrote:
Hello,
We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the SIP CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present.
Does Freeswitch fullfills all the recomendations there?
Thanks, __Yehavi:
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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. :-\ Steve On 03/22/2011 12:07 PM, Michael Collins wrote:
No, we have not paid them thousands of dollars for them to say that we are "SIP CONNECT Compliant." That being said, I read through some of the 1.1 specs. I didn't see anything that was egregious, but I also did not look at all 45 pages. This much I can tell you: to be "SIP CONNECT Compliant" you would need to program your dialplan for some specifics. Here's an example:
Section 13 of the spec says:
*13 Emergency Services*
The SIP-PBX MUST have a dial plan that recognizes emergency calls.
This will differ from one country to the next and is a function of the administrator's dialplan configuration. There are no inherent limitations in FreeSWITCH to prevent you from being up to spec on this. However, out of the box with a default installation you won't have this already configured.
My guess is that we could write tests that demonstrate that FreeSWITCH *could* be SIP CONNECT compliant, however I doubt that would be of much value because you have to cough up a minimum of $2500 to have their seal of approval. Beyond that look at their membership list:
http://www.sipforum.org/component/option,com_fullmember/Itemid,195/
It's a who's who of retarded SIP providers and PBXes. They've got Sonus on their list for cying out loud! That alone should invalidate the entire SIP CONNECT venture IMHO.
So, I would answer your supplier this way: As far as I know there aren't any limitations in FreeSWITCH that would prevent it from complying with the SIP CONNECT recommendations.
Oh, and GLWT! :)
-MC
On Mon, Mar 21, 2011 at 11:46 AM, Yehavi Bourvine <yehavi.bourvine@gmail.com <mailto:yehavi.bourvine@gmail.com>> wrote:
Hello, We published an RFP for our SIP connection to the PSTN. One of the suppliers asked whether Freeswitch complies with the SIP CONNECT recomendation. I have to answer them soon, and don't have enough time to read and evaluate the entire document at present. Does Freeswitch fullfills all the recomendations there? 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
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
_______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
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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-- Kristian Kielhofner
participants (5)
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curriegrad2004
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Kristian Kielhofner
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Michael Collins
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Steve Underwood
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Yehavi Bourvine