Okay, need some feedback on my perception of transcoding and how I plan to implement.
- Default codec for all phones connected to a cluster is g711
- One site (site B) is remote and has limited bandwidth WAN bandwidth to the main site (site A). It is recieving calls from PSTN via the main site's PRIs
- Phones at sites A and B are in their own device pools with their own regions configured.
I would like to configured the each site to use g711 locally and g729 remotely. So, this is easy to do via region configuration and I have accomplished it with no problems - when calls are established between phones at different sites g729 is being used.
Now, to add the complexity - the remote site's main number goes to a call handler on the main site's CRS that allows a caller to select options to reach people at the remote site. When the call handler is accessed internally, no problem works fine. When the call is coming in through the gateway, it goes to the CRS call handler which presents the options. When an option is pressed and the call is connected - no sound. So, in this case where a call is happening inbound from the main site's PSTN circuit to the remote site's endpoint, the two endpoints in this case are the gateway and the phone at the remote site. The gateway, by the way, is part of the site A's Device Pool/Region. So, in my experience when there is a codec mismatch there should be a reoder signal when the endpoints try to connect, not a mute sound. This seems odd to me. My theory is that the gateway needs to be configured to connect calls to the remote site using G729. Is the transcoding configured locally on the gateway then?
The codec will be renegotiated in accordance with the region configuration when CCX tells UCM to redirect/transfer the call to the phone at site B. As long as the phone supports G.729a or ab a transcoder should not be required for this.
Silence usually indicates that RTP is being lost. Press the help button (?) twice on the phone and see if it is receiving RTP packets from the gateway.
Is the audio loss one way (which direction?) or bidirectional?
Are there any ACLs or firewalls between the PSTN gateway and the phone?
If you call the DID on this phone directly, is audio lost then as well?
Edit: There was also an excellent reply to a similar post that suggests there may be a few bugs if your gateway is using MGCP:
Yeah, it seemed odd that the symptom was loss of audio instead of a reorder. More unusual that it would only happen after adjusting the regions for G729 and not happen with G711. However, this bug that you mentioned helps explain it.
To answer your question, the loss of audio is in both directions. No ACLs, and there is audio loss when calls are made directly from the gateway as well instead of being handled via CUCCX.
Will take a look at the bug mentioned in your post, this is in fact an MGCP gateway so I may be having the problem with that bug.
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