I have a network with two sites, A and B. Site A is where the CallManager servers, PSTN gateway, etc. reside. Site B only has IP phones.
I specified a locations bandwidth of 160k for site B and unlimited for site A. All devices, media resources, etc. are configured with the correct location and are using G.711.
The scenario I'm running into problems with is conferencing between these sites. Here's one case: A user in site A calls a user in site B. The user in site B tries to conference a PSTN user but when they try to complete the conference they get the message on their phone "cannot complete conference". If I bump the location bandwidth to 240k, it works fine.
The part I don't understand is why I need to configure any more than 160k for this scenario. If each audio stream is 80k, then the locations bandwidth is totally used up during the conferencing process by the A-to-B call plus the B-to-PSTN call. However, as soon as the conference is completed (after the second "Confrn" button press), there is only one stream going to site B, which uses 80k.
Having this work when the location is configured for 240k implies that there are three streams passing between sites A and B during the conferencing process, which I'm not sure how that's possible. It should go from a max of 2 streams down to 1 when the conference is complete.
I don't want to leave 240k configured because that's more bandwidth than I want to use over this path but I'm not sure what else to do.
Yes, I know that I could configure G.729 over the WAN and engage transcoders at site A to do conferencing but in addition to really trying to understand what's going on here, I want to use G.711.
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