[Asterisk-Users] Several H323 bugfixes - working SIP <-> H.323 translator

[Marian Durkovic]

On Fri, Mar 19, 2004 at 11:25:59AM +0100, Paul Cheng wrote:
> Hi,
> 
> The patches also did not help us and in fact created some new problems.  
> The old chan_h323 could pass on early audio and provider messages, but  
> after the patch, this capability is gone and the channel only rings and  
> rings while the provider is sending the message.

I've not removed the early audio cut-through.

For SIP->H.323 direction, the patched version sends:

180 Ringing           when Alerting PDU is received from H.323 side
183 Session Progress  when asterisk starts getting RTP packets
                     (this is handled in chan_sip.c regardless of H.323 state
                      and I left it untouched)

Calls with inband info get both of them i.e. 180 first and 183 afterwards.
This might perhaps confuse some SIP clients, but is legal according
to RFC3261 and works fine e.g. with Cisco 7940s or Xlite. I'll appreciate any
info if this is the problem.

The original version never sends 180 Ringing due to various bugs. Thus the
SIP caller gets no ringback tone for H.323 calls without inband info.


For H.323->SIP direction, the patched version sends:

Alerting PDU            when 180 Ringing received
Progress PDU with PI=8  when 183 Session Progress received

The original version doesn't detect SIP states and it sends Alerting PDU
immediately (even if the user does not exist or is busy).



	With kind regards,


		M.

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----   Marian Durkovic                       network  manager         ----
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----   Slovak Technical University           Tel: +421 2 524 51 301   ----
----   Computer Centre, Nam. Slobody 17      Fax: +421 2 524 94 351   ----
----   812 43 Bratislava, Slovak Republic    E-mail/sip: md at bts.sk    ----
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