Jade, RFC 2833 cannot insure delivery--there are no acks and therefore no retransmissions. It does, however, rely on a certain amount of redundancy so that if packet loss is low enough for voice, it is probably low enough for DTMF.
For example, since it looks like you are using G.723.1, I assume your packet interval is 30ms. Therefore, at the beginning of the DTMF signal, stop generating audio packets and start generating DTMF packets at the same interval--every 30ms. While the signal continues, continue generating DTMF packets every 30ms. When the signal ends, transmit the last DTMF packet three more times (for a total of four times--some implementations get this wrong) and then switch back to generating audio packets. It works pretty well. Paul -----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Jade Chen Yan Sent: Monday, February 17, 2003 9:26 PM To: [EMAIL PROTECTED] Cc: [EMAIL PROTECTED] Subject: [Sip-implementors] Re:DTMF in SIP Dear Paul, Thank u for your help.We inserted the DTMF into the RTP audio stream (G.7231). The packets r often lost. Many people recommend using RFC 2833 . But I wonder how RFC 2833 can insure the reliability transport of DTMF events.I cannot get much from the RFC 2833. Regards, Jade _________________________________________________________________ Original Message: Message: 1 From: "Paul Long" <[EMAIL PROTECTED]> To: "Sip-Implementors \(E-mail\)" <[EMAIL PROTECTED]> Subject: RE: [Sip-implementors] DTMF in SIP Date: Sat, 15 Feb 2003 18:57:54 -0600 Adetya, Use RFC 2833 (http://www.ietf.org/rfc/rfc2833.txt). In your original query, by, "we use RTP to carry the DTMF, but we found that it was not reliable," did you mean that you inserted DTMF into the RTP audio stream as audio or that you have implemented RFC 2833 but found some problem with it? If the latter, what was the problem? Paul _______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
