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1. H323 is a good protocol if you must translate ISDN or SS7 signaling 
into VoIP, this is why carriers love it. But it dosen't pass NAT well, is 
difficult to implement (right now because of OpenH323 - thanks to Craig 
and Robert, is not such a problem any more), which has lead to huge 
problems in the past. It lacks IM. Has a lot of implementations. 

the real VoIP protocol, is flexibile and powerfull. Carriers
don't use it much because it dosen't have equivelents for all commands and
codes for ISDN. The equipements are cheap and are becoming cheaper and
cheaper. The implementations (at least those in Linux) are rubbish, in
most cases lacks features, and a lot of them also lack stability (at least
those that i have tried). It has IM support including subscription,
notify. It also have a lot of problems on passing the NAT but is much
easier then H323, because is text not binary.

3. IAX is a not only a signalling protocol like H323 and SIP but also
carry media on the same port. Is good because is passing the NAT. But in
fact IAX has the same problems like Asterisk (IAX stand for Inter
Asterisk Exchange), is a PSTN PBX with support for VoIP (like in fact most
of the solutions). Now don't get me wrong. Because of that IAX is solving
the NAT problem. Even encrypted it will pass the NAT, which is great. It
dosen't have IM (it has text messages). In the same time you can't have
proxy just for signalling (like in H.323 or SIP), or just for RTP. For the
IM exist an extension made by Firefly but i didn't see any RFC. I must
addmit i didn't ask for one and Firefly company is willing to work with
other companies on that standard.

Regarding the security :
1. H.323 has H.245 encrypted tunnel, and i don't know anything about RTP. 
(maybe someone else can explain me what should happen here).
2. SIP can be encrypted with TLS and for RTP can use sRTP. I know a few 
implementations i have no idea how well are working.
3. IAX even if is the most easy to solve the encryption problem it didn't
had this feature until recently. There is no RFC or a clear idea about how
this should be done. 

Diana Cionoiu

3 May 2010:
Yate 3.0.0 alpha 3 released. Featuring the new Jabber server and wideband audio.
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8 March 2010:
Yate 2.2 released. Mostly bug fixes. Dahdi compatible. Latest 2 release before 3.0.

6-7 February 2010:
Yate booth at FOSDEM 2010. Free CD with Freesentral available.

2 Nov 2009:
Yate 2.1 launched. Can replace a Cisco PGW2200 to control a Cisco AS54xx.

6 Aug 2008:
Yate and OpenSIPS (former OpenSER) join to build IP based clusters.

4 Aug 2008:
Yate 2 launched.

10 Jul 2008:
Yate presentation in Germany.

Feb 2008:
Yate 2.0.0 alpha 2 released. New routing module allows sending ENUM routed or forked calls to numbers of registered phones. More...

21 Jan 2008:
Yate 2 alpha released. Major changes, new ISDN, SS7 and MGCP stack. Added analogic and RBS support.

3 September:
Yate 1.3 released. Minor fixes and improvments mainly in client and SIP.

14 August:
Yate based ISDN passive recording system released by Trisys.

16 April:
Yate 1.2 released. Added Jingle and XML support, PBX improved.

25 September:
YateAdmin 1 released.

25 September:
Yate 1.1 released. Fallback routing from a database, fax support in Linux and bug fixes. Changelog and Download availables.

11 July 2006:
O'Reilly published an article about prototyping telephony applications with Yate and Python.

10 July 2006:
Yate 1 released. Includes YIAX, YSIP, YRTP and many new features.

June 1st 2006:
New Yate website launched


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