[Asterisk-Users] * as a SoftSwitch/Router solution

[Nir Simionovich]

Hi All,

  I've been experimenting during this weekend with asterisk as a softswitch, 
talk about me being completely lifeless, but let not talk about that.

  I've been conducting some really funny tests, trying to get an optimal 
SoftSwitch functionality. Here is my current setup:

Source: Windows XP Pro + SJphone
Box 1: Asterisk running in PassThorugh mode
Box 2: Asterisk running in PassThrough and ZAP termination mode.
Box 3: VocalTec VG2000 H323 gateway

  Here are the 3 tests I've conducted:

1. SJphone/SIP -> Box1/SIP -> Box2/Zap -> PSTN -> My Mobile Phone
   That didn't work, which was caused by the fact that my SJphone is located 
behind a NAT gateway, which happens to be Box 1. (Silly me, no?)

2. SJPhone/SIP -> Box1/SIP -> Box1/IAX2 -> Box2/IAX2 -> Box2/Zap -> PSTN -> My 
Mobile Phone
  Worked nice, the IAX transcoded the codec to GSM, which worked lovely, voice
was clear and crispy. Works very nice!!!

3. SJPhone/SIP -> Box1/SIP -> Box1/IAX2 -> Box2/IAX2 -> Box2/H323 -> Box3/H323 
-> PSTN -> My Mobile Phone
  Worked half way only. Call was terminated nicely to the mobile phone, while 
the voice traversed from the mobile phone to the SJphone, the SJphone was 
unable to send the voice back to the mobile phone. I've noticed the 
following: the codec between SJphone and Box1 was aLaw (they are on the same 
LAN), Box1 to Box2 was GSM (format 4 I believe), Box2 to Box3 was aLaw again 
(The VocalTec box currently supports aLaw only). I think that either the 
somewhere along the way, the transcoding of one of the channels got jipped, 
or this crazy setup is too crazy to work (although, logic suggests that it 
should)

  Had anyone else conducted crazy tests like these? especially with 3rd party 
vendors? I would really like to know the outcomes.
-- 
Regards,
  Nir Simionovich
  nirs at net-gurus.net
  Net-Gurus.Net - Security by Design