Must the "telephone-event" have the same frequency as the codec used in the call?

Question

I use RFC2833 as the DTMF transmitting method for the calls.

Q1: Must the "telephone-event" have the same frequency as the codec used in the call? E.g. If I use SPEEX 16000 then can I have telephone-event/8000?

Q2: And can I have SDP without any audio codecs but, with specified "telephone-event"? E. g. can I have an SDP like that:

m=audio 12346 RTP/AVP 100
a=rtpmap:100 telephone-event/8000
a=fmtp:100 0-15,66,70

Answer 1

Interesting question, I am wondering what usage you have in mind.

First some standard related piece of information:

1. rfc4733 is applicable here. rfc4733 has obsoleted rfc2833.

2. In rfc4733, play-out in audio stream of telephone-events is described, it is recommended to use the same rate of the audio stream.

   So Q1 answer is positive, in theory you can have mixed rates. It means you do not follow the recommendation, you are on your own! In practice, only equipment supporting multiple rates would accept it.

Q2 is not really clear I guess. If it is a special case of Q1 with no audio stream, I doubt that use-case is supported at all. After all, both share the same RTP SSRC field.

Answer 2

Q1: Yes. Here is the proof, taken from RFC errata:

Named telephone events are carried as part of the audio stream and if they use the same SSRC (therefore the same timing and sequence number space), they MUST use the same timestamp clock rate as the regular audio channel.

Q2: Most probably, yes. But, still, I'm not very sure.