List users, A recent post on MeetMe timing mentioned the internal_timing option, which can be configured to have Asterisk asynchronously generate outgoing RTP when a timing device (ie. ztdummy) is available. This allows Asterisk to produce outgoing audio in situations where no incoming audio is arriving due to, for example, silence suppression.
This seems like a major improvement to the core functionality of Asterisk, but a search of the lists, the wiki, and the book didn't produce much information about the option. We are a Business Edition shop running ABE-B, which is based on the 1.2 codebase. However, the 1.4-based ABE-C was recently released so I'm interested in the potential pros and cons of internal_timing. For instance, how much does turning on silence suppression on a phone typically lower the bandwidth requirements per call and what are the effects on features such as VAD and CNG? Regards, Matthew Roth InterMedia Marketing Solutions Software Engineer and Systems Developer PS - I apologize for the threadjack, but it seems that my posts never make it to the list when I try to start a new thread. _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
