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SIP session border controller


Like we have seen in the previous chapter Yate implements the Session Initiation Protocol for communication among others. Besides the SIP router functionality, Yate also can be used like a SIP border controller.

You may wonder what exactly is a border controller?. This is what we will describe in this chapter.

A typical session border controller is used to overcome some of the problems that may appear in a VoIP communication by integrating it between the caller and the called party signaling path. Such problems can be of various types. The more important ones are like it follows:

The traversal of signal and media across a NAT or firewall border device can fail for various reasons. Letís take them piece-by-piece. First, there is a firewall problem, even it doesnít apply every time and for every firewall. It is about the probability that the dynamically opening and closing of ports on incoming calls might just not work. The main problem is the Network Address Translation (NAT) one. Here is about the IP addresses that are inserted in the packets by the VoIP clients, which addresses belong to the private network behind the NAT server, and consequently canít be routed outside it. The session border controller is positioned between the clients with these problems and solves them so that the traffic can be possible. There arenít only the signaling packets the ones that are passing through the border controller but the media ones too.

Another reason to use a session border controller is due to different service providers for the clients, which states for a connection usually passing through many different codec hops. Here fits the border controller as a cheaper way to provide media translation and in addition also accounting information and security services.

A use for a session border controller is also to collect usage information for each session. This can be used to create Charge Detail Records (CDRs) based on various factors as media type, bandwidth used, account information, etc.

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 Ė 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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