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H323



Mizu VoIP-GSM Gateways can accept SIP and H323 registrations, can act as a SIP proxy or a H323 Gatekeeper or Gateway. These functions can be run simultaneously.


The mizu voip server is capable to handle h323-h323 wholesale or transparent sip to h323 or h323 to sip conversion.
Both H323 (v.1,2,3,4) Gateway and Gatekeeper mode are enabled and protocol variations are handled automatically (RAS, H.245, H.225, media proxy or bypass, Fast Connect/Fast Start)
H323 should be used only for gateways. For endusers you should always prefer SIP over H323 because:
  -some class 5 features  will work only with SIP protocol
  -H323 GK doesn’t support username/password authentication (only ip/port and/or techprefix based)

Files needed (part of the install package):
atarongk.exe -main h323 gateway and gatekeeper
vsip.exe -for SIP to H323 conversion (will not be used for H323 to H323 calls)
openh323.dll, ptlib.dll, pwlib.dll, libyate.dll -required dll's
sipcfg directory and sipunits directory -required for sip to h323 converter (vsip.exe)

To enable h323 the hash323 and can_h323 global config values must be set to true.
To enable h323-SIP conversion the runsipproxy global config values must be set to true.
These are can be enabled from the Configuration Wizard.
The atarongk application can be controlled from the console port if you type the “gk” command

Two important user entry is created automatically during setup used internally for sip to h323 conversion (and inverse):
  -"sip2h323caller" traffic sender
  -"sip2h323" sip proxy
Make sure that these are present and enabled.

To setup a h323 caller (incoming traffic), there is no special action to do. You just have to add a traffic sender user with the proper authentication.
For outgoing traffic you must use a "H323 GW/GK" user type and then add it to your routing

In case of H323 entries, the "callsigaddress" field is used instead of the "port" field. This must be set to 1720 usually (standard H323 signaling port)
The media ports are negotiated automatically.

Technical backgrounds:
h323 to h323
  1. incoming call arrive to atarongk
  2. atarongk will call the routing and will get the destination address
  3. atarongk forwards the call to the destination
h323 to sip conversion:
  1. incoming call arrive to atarongk (h323 protocol)
  2. atarongk will call the routing
  3. call will be forwarded to vsip converter
  4. vsip will call the routing again (sip protocol)
  5. the rest of the call is handled by mizu sipstack
sip to h323 conversion:
  1. incoming call arrive to mizu sipstack (sip protocol)
  2. call will be sent for vsip
  3. vsip will send the call to atarongk (h323 protocol)
  4. atarongk will call the routing again
  5. atarongk will forward the call to the destination

Troubleshooting:
  1. replace the atarongk file with the _dbg version
  2. change the "sipcommand" global config option to "vsip -vvvvvvvv -l siplog.dat"
  3. change the "gkcommand" global config option to "atarongk -ttttt -o gklog.txt"
  4. set loglevel to 5
  5. check the callsigaddress field for h323 gateways (usually 1720)
  6. start the service
  7. make sure that atarongk and vsip processes are running
  8. make a test call and check the logs
  9. for h323 test calls you can use the openphone.exe or the ohphone.exe (example: ohphone -n -p 192.168.1.7 -ttttt 2222)

   Download the Mizu H.323 test suite from here
 



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