RUOFF, LARS (LARS)** CTR **
2011-11-18 08:22:05 UTC
Hi all,
New to this list. Apologies if this question has been asked many times before (my searches showed that it has, but i was still unable to find a definitive answer).
The question is about SDP telephone-event (DTMF) payload negotiation.
Imagine the following call setup between A and B:
INVITE A->B
SDP:
(among other media formats)
a=sendrecv
a=rtpmap:101 telephone-event/8000
200 OK B->A
SDP:
(among other media formats)
a=sendrecv
a=rtpmap:97 telephone-event/8000
The question is:
Is the above legal? I.e. is B allowed to choose a different PT than proposed y A.
In case yes, what PT should the telephone-events be sent...
from A to B?
from B to A?
Please corroborate your answers by providing normative references if possible.
I studied RFC 3264 (SDP Offer/Answer Model) intensively, but could not conclude for a definitive answer in that particular case.
Regards,
Lars Ruoff
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Use sip-***@cs.columbia.edu for questions on how to develop a SIP implementation.
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Use ***@ietf.org for issues related to maintenance of the core SIP specifications.
New to this list. Apologies if this question has been asked many times before (my searches showed that it has, but i was still unable to find a definitive answer).
The question is about SDP telephone-event (DTMF) payload negotiation.
Imagine the following call setup between A and B:
INVITE A->B
SDP:
(among other media formats)
a=sendrecv
a=rtpmap:101 telephone-event/8000
200 OK B->A
SDP:
(among other media formats)
a=sendrecv
a=rtpmap:97 telephone-event/8000
The question is:
Is the above legal? I.e. is B allowed to choose a different PT than proposed y A.
In case yes, what PT should the telephone-events be sent...
from A to B?
from B to A?
Please corroborate your answers by providing normative references if possible.
I studied RFC 3264 (SDP Offer/Answer Model) intensively, but could not conclude for a definitive answer in that particular case.
Regards,
Lars Ruoff
_______________________________________________
Sip mailing list https://www.ietf.org/mailman/listinfo/sip
This list is essentially closed and only used for finishing old business.
Use sip-***@cs.columbia.edu for questions on how to develop a SIP implementation.
Use ***@ietf.org for new developments on the application of sip.
Use ***@ietf.org for issues related to maintenance of the core SIP specifications.