[Freeswitch-users] multiple leg and multiple rtp

[Sergey Okhapkin]

Critical issues are when SIP error come after 18X provisional response.

- if bypass_media is false then dialplan stops and leg a is explicitly hang up 
(switch_ivr_bridge.c, line 513).
- if bypass_media is true, then dialplan continue, but there is 10 seconds 
delay before next bridge application sends INVITE to gateway ( 
http://lists.freeswitch.org/pipermail/freeswitch-users/2010-January/024354.html ) 
I didn't track down yet why this happens looking at FS sources.

Why I didn't open a bug on jira? Because FS behaves according to the design 
and specs :-) But not according to real world requirements... 


On Wednesday 13 January 2010, Brian West wrote:
> Can you elaborate on these "Critical" issues you seem to be having?  Why
> aren't you opening a jira for them if they are that critical to your needs?
>
> /b
>
> On Jan 13, 2010, at 9:29 PM, Sergey Okhapkin wrote:
> > If you wish FreeSWITCH to be well adopted by the community, then FS
> > should follow the real world "rules" but not specs...
> >
> > Don't you agree that 99% of SIP servers are set up to interconnect
> > with "buggy" PSTN and should follow PSTN rules, but not SIP specs?
> >
> > Some background - I did run asterisk for years, but switched to FS
> > recently because of critical asterisk problems with SIP handling when
> > asterisk is not in the media path. I spend a lot of time porting the
> > billing system to FS. And I did it. What I got? Critical problems with FS
> > early media handling. Hopefully I can switch back to asterisk if FS
> > problems with early media will begin to draw customers away.
>
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