Hi All, So I've been exploring the use of G.722 encoded wideband audio recently. I have three different SIP devices that allow this: Eyebeam, IP650 and a Siemens S865IP. The Siemens and IP650 seems to work fine together. Calls pass between them in what the Polycom notes as "HD" mode and the audio quality is certainly very good.
However, things are not so easy with Eyebeam and the IP650. When a call is placed between those two the audio stream from Eyebeam to the IP650 is never heard. The audio from the IP650 to Eyebeam is heard, and very good quality. David Frankel of ZipDX tells me that there is an error in RFC3551 such that G.722 RTP clock/timestamps are actually wrong. To quote the RFC directly. "Even though the actual sampling rate for G.722 audio is 16,000 Hz, the RTP clock rate for the G722 payload format is 8,000 Hz because that value was erroneously assigned in RFC 1890 and must remain unchanged for backward compatibility. The octet rate or sample-pair rate is 8,000 Hz." It seems that some manufacturers adhere strictly to the RFC while others correct for the error. As such there are problems with G.722 interoperability. Counterpath defends their implementation as being according to the RFC. This begs the question of what does Asterisk do with G.722? I've yet to try v1.6 so I open the question to the group. Michael -- Michael Graves mgraves<at>mstvp.com http://blog.mgraves.org o713-861-4005 c713-201-1262 sip:[EMAIL PROTECTED] skype mjgraves fwd 54245 _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
