Hello, I am working with several SIP projects that use g722, or are trying to do so, with pjsip library.
According to pjsip team's interpretation of g722, it works with 14bits PCM for input/output, so pjsip basically 'converts' the audio sample from 16 bits to 14 when encoding and vice-versa. Some implementations don't do 16<->14 bits conversion, so when pjmedia talks to one of those the over-driven audio problems appear. What we need to know is what's the most used implementation: 14<->16 bits conversion or not. Any pointers to help clear this up? We'd really like to see more g722-capable SIP clients for our own conference on ZipDX. Regards, Randy http://vuc.me _______________________________________________ -- 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
