"Cary Fitch" <[email protected]> writes: > Is there a plain 64K codec that would simply pass through the SIP server and > be handed off to a PRI or phone co. trunk on a T1 on the other side of the > SIP server? Digital 64K telco sounds very good as a phone conversation.
You can't get a guaranteed bit-for-bit identical stream through SIP/RTP or IAX. You can pick the same codecs as the PSTN uses (Alaw or ulaw, depending on country), but jitter and packet loss still makes things like DTMF or fax/modem unreliable. For DTMF it is better to signal that in RTP or SIP, for fax you want T.38, and for modems you need incense and strange rites at midnight. /Benny _______________________________________________ -- 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
