[Freeswitch-users] Equivalent to Asterisk's "directrtpsetup=yes"?

[Kristian Kielhofner]

On Wed, Jan 27, 2010 at 7:38 AM, Fred-145 <codecomplete at free.fr> wrote:
>
> Hello
>
> Thanks to "directrtpsetup=yes", it appears that Asterisk can act as an SIP
> proxy and a PBX at the same time, ie. RTP packets will flow directly between
> the two SIP end-points with Asterisk still being able to provide PBX
> services like call transfer, MoH, etc.
> The point is to lower latency for packets to reach their final destination
> and lower CPU load on the Freeswitch server.
>
> Does Freeswitch offer the same feature, or must RTP packets always go
> through the Freeswitch servers?

  This isn't exactly true...  If you do some research you'll find that
OEJ (the author of the Asterisk SIP channel driver) does NOT recommend
the use of directrtpsetup because its use hasn't been tested with many
scenarios (including some you describe, I'm sure).  AFAIK it's still
marked "experimental".  The last time this came up on Asterisk-users
here was the exchange:

Kevin P. Fleming wrote:
> Kristian Kielhofner wrote:
>
>
>> What version of Asterisk is this?  Last I heard (from Olle) this
>> option was very experimental and should not be used on production
>> systems.
>>
>
> He even helpfully documented it that way in the sip.conf.sample file,
> along with a list of (known) cases where it will fail, although there
> are probably plenty more.
>
>

  So Kevin Fleming agrees.

  Needless to say FreeSWITCH has bypass_media.  Last I heard
FreeSWITCH will re-INVITE itself back in the media path if you put a
call on hold (for example) but it won't go back to bypass_media when
you take the call off hold.  Anthony said it would probably take about
$500 in bounty to get that functionality.  Then again, maybe he just
decided to do anyway it because he thought it was cool.  That's been
known to happen too.

-- 
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
http://www.submityoursip.com
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