Main • SIPRouter

SIP Router

First of all let’s see more exactly what SIP means.
SIP comes from Session Initiation Protocol. The Session Initiation Protocol is a signaling protocol developed and standardized by the Internet Enginneering Task Force (IETF).

What does this definition mean in our case? This means it is used to initiate and maintain sessions in a network to provide VoIP communication between two or more points. Such points could be one phone and one pc for example or more computers located in various places in case of a multi-conference, phisycaly connected in such a way that VoIP communication to be possible.

There are various protocols used to carry data over a real time multimedia session in numerous formats like voice, video or text messages. SIP works along with them, helping the endpoints in a communication to agree about the session characteristics. SIP doesn’t offer any other services besides the session configuration related ones, that is why is being used along with protocols specialized on other tasks like the Real-time Transport Protocol (RTP) for data transporting and Quality of Service, the Session Description Protocol (SDP) for describing multimedia sessions and others. This way SIP can provide complete functionality to the users but despite all this collaboration with other protocols it does not depend on them.

As a SIP router Yate supports this protocol and its features like registering, proxy and redirect, through its own library that implements them, this being called YASS – Yet Another SIP Stack. In the following lines are described some more features that an implementation of SIP permits.

Besides establishing, modifying and finalizing a session, SIP can also be used to invite participants to an already existing one. Also a type of media can be added or removed from a session. Like stated in the RFC document 3261 which represents the last published standard for SIP, the protocol supports name mapping and redirection services and users can maintain a single externally visible identifier regardless of their network location. The call forwarding SIP provides to the implementing servers is accompanied by the possibility of negociating the terminal type and capabilities and selecting it. This way a caller is given a choice about how to reach the party, via Internet telephony, an answering service, etc.

The security of communication services always was considered important so SIP provides a suite of security services like user authentication, denial-of-service prevention, integrity protection, and others.

SIP addresses users using an e-mail like easy-to-understand addressing system. If you are interested in a more technical approach of this and the way SIP is used in a redirect and a proxy mode feel free to consult the schemes present in the Appendixes.

July 2014:
Yate 5.4 and YateBTS 4 launched. Added JSON and DNS support in Javascript, Handover support in YateBTS.

March 2014:
YateBTS 2.0 launched. Added authentication and WebGUI. Added USSD support in commercial version.

March 2014:
Yate 5.2 launched. Better JavaScript support and a fixed memory leak.

Jan 2014:
YateBTS 1.0 launched. The first GSM Basestation which works with an IMS/VoLTE core network.

Jan 2014:
Yate 5.1 launched. Better JavaScript support and added libygsm. Elisa chatbot added in RManager

Oct 2013:
OpenHSS is the Yate based HLR/HSS solution for MVNO and LTE carriers.

Oct 2013:
Yate 5 released. Added IPv6 support in SIP for LTE. Improved JavaScript support. Download NOW

Jan 2013:
Yate 4.3 released: Added XML support in Javascript. SCCP - GTT routing between different networks. Stability improvements.
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Aug 2012:
Yate 4.2 released: SIP flood protection. Better Jabber/Google Voice support. Usable Javascript. Fixed SIGTRAN links fluctuations.
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Apr 2012:
YateClient was accepted in the Mac Store.

Yate 4.1 released: better Gvoice support, iSAC codec, support for new Wanpipe drivers. Fixes – T.38 and Mac client issues.

Mar 2012:
SS7Cloud is launched today, 1st March, 2012, by NullTeam, Yate creators. Having all you need to be a US CLEC, it brings SS7 services in a cloud.

Feb 2012:
Yate 4.0 released.
SCCP, TCAP, MAP and CAMEL, TCP and TLS in SIP, Javascript fast prototyping of telephony applications and brand new face for YateClient.

Nov 2011:
Here is a video that, quote "demonstrates the truly awesome power of the YATE engine, as it easily handles 3 simultaneous calls to an audio player application including dtmf (button press) handling "(from PaintedRockComm).

Nov 2011:
Yate will attend ORR - OPENRHEINRUHR (November 12 - 13).

04 May 2011:
sipgate chooses open source project Yate for core infrastructure.

12 Apr 2011:
Yate 3.3.2 released.
Fix for Jingle calls to Google Voice dropping after 5 minutes.
4 Apr 2011:
Yate 3.3 released.
Support for GMail chat conference, fixes for internal microphone in MacOS. Minor fixes in SS7 M2PA and ANSI. Fixes in H.323, SIP and RTP.

9 Mar 2011:
Yate 3.2 released.
Bug fixes in SIGTRAN/MGCP/SS7 and added support for CNAM/LNP lookup by SIP INVITE/3xx.

Feb 2011:
Yate will attend FOSDEM and XMPP summit.

31 Jan 2011:
Yate 3.1 released.
Yate client support for Google Voice. Support for any country tones in tonegen.

20 Dec 2010:
Yate 3.0 released.
SS7 ITU certified. SS7 STP added. Client supports Jabber IM (Google Talk + Facebook).

3 May 2010:
Yate 3.0.0 alpha 3 released. Featuring the new Jabber server and wideband audio.

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.

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