Asterisk as a softswitch in a IP-Centrex environment?

Archived from groups: comp.dcom.voice-over-ip (More info?)

Hi all!

We want to develop an IP-Centrex solution based on a softswitch
architecture and want to hear from VoIP community their experiences,
since Asterisk seems to be a good candidate for the job.

Basics of the architecture are:

* SIP IP Phones (who will be using the IP-Centrex services) connected
through a managed IP network to the softswicth;
* Softswitch connect through LAN to a signaling/media gateway, using a
proprietary signaling/media gateway control protocol (no MGCP/H.248
for now, unfortunately);
* Signaling/media gateway connected to the PSTN using SS7 on the TDM
side;
* Signaling gateway is SIP capable on the IP side;
* Media gateway is capable of: codecs (of course!), conferencing, DTMF
tones detection/generation, message recording/playback using an
external NFS server as a media server;

For a nice diagram of this kind of architecture, please take a look
at: http://www.ip-centrex.org/how/softswitch.html.

Some numbers to give an idea of the "size" of the system:

* Maximum number of IP-Centrex clients: 5,000;
* Signaling/media gateway connection to the PSTN: DS3 (672 DS0's);
* Signaling/media gateway DSP "resources": 1,024 (that can be
configured for codecs, conferencing, DTMF tones detection/generation,
message recording/playback);

We want Asterisk to:

1. If the caller and the called parties are both SIP clients, route
the packetized voice streams directly to one another; consequently the
voice stream never reaches the Asterisk box (or boxes);
2. If the called party is served by the PSTN, instruct the originating
IP Phone to route the packetized voice stream the media gateway;
3. Based on (1.) above, use the media gateway DSPs conferencing
capabilities in the conference feature;
4. Based on (1.) above, use the media gateway DSPs message
recording/playback capabilities in the voice-mail and IVR features;

We understand the we may have to:

* Develop of an Asterisk channel interface so the Asterisk can control
the signaling/media gateway;
* Adapt Asterisk conference, voice-mail, IRV to achieve (3.) and (4.)
above;

I'm looking forward to hear from the VoIP community their
thoughts/comments on the Asterisk use in such a development:

* How Asterisk adapts to such number of IP clients (5,000)?
* Has Asterisk been used to control a signaling/media gateway in the
way we are planning to do?
* How hard it is for Asterisk to achieve requirements (1.) to (4.)
above?

Besides that, what are your experiences with Digium commercial
Asterisk support, especially in a project with such size/requirements?

Thanks in advance.

André Chrcanovic
FITec Inovações Tecnológicas
achrcanovic@fitec.org.br
Tel: +55 31 3263-4016