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A Media Gateway (MGW) acts as a translation unit between disparate telecommunications networks such as PSTN, Next Generation Networks, 2G, 2.5G and 3G radio access networks or PBX. Media Gateways enable multimedia communications across Next Generation Networks over multiple transport protocols such as ATM and IP.
Because the MGW connects different types of networks, one of its main functions is to convert between the different transmission and coding techniques. Media streaming functions such as echo cancellation, DTMF, and tone sender are also located in the MGW.
Media Gateways are controlled by a Media Gateway Controller (MGC) or Softswitch which provides the call control and signaling functionality. In the context of IMS however, the term "Softswitch" is rarely used.
The simplest open-source method of creating your own MGW is to use Asterisk. Asterisk is an open source VoIP PBX Telephony platform that supports SIP - the fundamental protocol used by IMS. An Asterisk MGW will essentially allow SIP/RTP based calls from an IMS network to breakout to the PSTN.
For this to work, some sort of ISDN card (with driver) is required to be installed on the same server in order to receive/make PSTN calls. Any HFC based ISDN card should work really well. You will need to install the correct kernel module driver -> Libpri and Zaptel. Get more info here.
As with all zaptel drivers you can have any amount of these cards in your Asterisk box, only limted by the amount of PCI ports available in your system.
Below are a few lines for sip.conf and extensions.conf to get you started on Asterisk:
sip.conf
[general]
context=ims-outgoing ; Default context for incoming calls
allowguest=yes ; Allow guest calls
extensions.conf
[ims-outgoing]
;Use zaptel to dial the phone number contained in the SIP INVITE URI
exten => _X.,1,Dial(Zap/1/${EXTEN})
exten => _X.,2,Hangup
[from-pstn]
; Answer PSTN calls with a default recording
exten => _X.,1,Wait(2)
exten => _X.,2,Answer
exten => _X.,3,Playback(tt-monkeys)
exten => _X.,4,Hangup
3G Video to SIP calls
This library from Asterisk Video Resources will allow Asterisk to bridge calls between a UMTS 3G mobile phone and a IP video phone (SIP or H323). The only hardware needed is one ISDN interface card (bri or pri) of the ones currently avaiable for Asterisk. Development is still in a very early stage, but the source code is avaiable in the subversion repository.
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