[asterisk-users] Asterisk B2BUA and Site to Site transfers

[Chris Bennett]

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