asterisk users - Dec 2007 - Asterisk B2BUA and Site to Site transfers

Hi All,

I am seeking input from anyone who may have seen a similar
configuration and dealt with similar issues to what I'm experiencing.

Configuration:
- 2 sites (site A and B)
- Asterisk 1.2.23 on each site (Trixbox)
- Internet 512/512 symmetric at each site, dedicated to VOIP calls
  only.
- IAX trunk between the sites, with data travelling across the 512/512
  Symmetric link
- PSTN inbound/outbound via a Sangoma PCI FXO card.

The required configuration is inbound calls at either site need to be
answered by a reception at Site A.

Calls coming in via PSTN to Site B, will result in a SIP extension at Site
A to be dialled and answered.  This will result in an active channel
between site B's asterisk server, and the user at Site A.

If Site A transfers that call *back* to site B, this will result in
another call leg being established to the user at site B.

Every RTP packet will travel:
- in via PSTN @ Site B
- across 512/512 DSL link to Site A's asterisk server
- back across 512/512 DSL link to user at Site B

We are noticing jitter and voice quality problems.  A call can degrade
in quality over time.  We are using G729 for the voice codec.  Can
anyone suggest further debugging I can do to determine the cause of
voice quality degradation?  Is there a way I can configure the
asterisk servers to not communicate the RTP traffic across the DSL
links and back again?

Any suggestions will be much appreciated.

Regards,

Chris Bennett

Chris Bennett wrote:
> Hi All,
> 
> I am seeking input from anyone who may have seen a similar
> configuration and dealt with similar issues to what I'm experiencing.
> 
> Configuration:
> - 2 sites (site A and B)
> - Asterisk 1.2.23 on each site (Trixbox)
> - Internet 512/512 symmetric at each site, dedicated to VOIP calls
>   only.
> - IAX trunk between the sites, with data travelling across the 512/512
>   Symmetric link
> - PSTN inbound/outbound via a Sangoma PCI FXO card.
> 
> The required configuration is inbound calls at either site need to be
> answered by a reception at Site A.
> 
> Calls coming in via PSTN to Site B, will result in a SIP extension at Site
> A to be dialled and answered.  This will result in an active channel
> between site B's asterisk server, and the user at Site A.
> 
> If Site A transfers that call *back* to site B, this will result in
> another call leg being established to the user at site B.
> 
> Every RTP packet will travel:
> - in via PSTN @ Site B
> - across 512/512 DSL link to Site A's asterisk server
> - back across 512/512 DSL link to user at Site B
> 
> We are noticing jitter and voice quality problems.  A call can degrade
> in quality over time.  We are using G729 for the voice codec.  Can
> anyone suggest further debugging I can do to determine the cause of
> voice quality degradation?  Is there a way I can configure the
> asterisk servers to not communicate the RTP traffic across the DSL
> links and back again?
> 
> Any suggestions will be much appreciated.
> 
I'm don't think setting reinvites on will fix your problem. The only 
thing I can think of is that you use some sort of call parking to park 
the call on SiteB's asterisk server and then have the person at siteB 
pick up the call from the parking lot

Anyone else know a better way to do this?