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@net-gurus.net
Net-Gurus.Net - Security by Design
