Thanks Ranjit,

The RFC also says

   A source MAY send events and coded audio packets for the same time
   instants, using events as the redundant encoding for the audio
   stream, or it MAY block outgoing audio while event tones are active
   and only send named events as both the primary and redundant
   encodings.

But the RFC doesn't address SIP/SDP coding for either technique.

I'm guessing that one would use this form

m=audio 8000 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000

when blocking audio, and this form

m=audio 8000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=audio 8002 RTP/AVP 97
a=rtpmap:97 telephone-event/8000

when sending both events and audio (on different streams).
I don't know how you would negotiate RFC2198 style packets
with both types of content.

Can anyone confirm or correct these assumptions?

-troy





[EMAIL PROTECTED] wrote:

Hi
   The RFC says "The Internet Telephony gateway detects DTMF on incoming
side and sends RTP packets instead of regular audio packets."

 So the Gateway can negotiate as supporting telephone-event in SDP as:

 m=audio 8000 RTP/AVP xyz
 a=rtpmap:xyz telephone-event/8000

 where 'xyz' is the dynamic payload type, since 'telephone-event' does
not have static payload type defined.


Regards Ranjit


-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Troy Cauble Sent: Tuesday, June 15, 2004 4:33 AM To: [EMAIL PROTECTED] Subject: [Sip-implementors] RFC 2833 SIP/SDP negotiation questions



RFC 2833 states that telephone-event RTP packets
can be sent instead-of *OR* in-addition-to voice.

Is it correct that negotiating telephone-event as
one format in a media stream

m=audio 8000 RTP/AVP 0 97
a=rtpmap:0 PCMU/8000
a=rtpmap:97 telephone-event/8000

implies the former ("instead-of"), while multiple media streams

m=audio 8000 RTP/AVP 0
a=rtpmap:0 PCMU/8000
m=audio 8002 RTP/AVP 97
a=rtpmap:97 telephone-event/8000

implies the latter ("in-addition-to") ??


Which technique is usually used?


In the latter case, would you expect the 8002 stream to send anything when no telephone-events were happening (normal voice)?


Thanks, -troy


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