[Date Prev][Date Next][Thread Prev][Thread Next][Date Index][Thread Index]
[Sipping] Question wrt transcoding
[I am new to the list, sorry if question is already answered]
Hi,
I have a (basic) question regarding O/A in the transcoding scenario; it refers to http://www.ietf.org/internet-drafts/draft-ietf-sipping-transc-3pcc-02.txt
If an answerer B can receive audio but cannot understand any of the codecs that the offerer A includes in the INVITE, it usually sets the port=0 in the corresponding m line in the answer. This is understood. But how does the SIP UA acting as controller know that this means "No, I don't want that m line" OR "I want it but I cannot support it"??
Is it by including an additional m line with the proposed formats in the answer as mentioned in RFC 3264:
"Rosenberg & Schulzrinne Standards Track [Page 9]
RFC 3264 An Offer/Answer Model Session Description Protocol June 2002
For streams marked as recvonly in the answer, the "m=" line MUST
contain at least one media format the answerer is willing to receive
--> with from amongst those listed in the offer. The stream MAY indicate
--> additional media formats, not listed in the corresponding stream in
--> the offer, that the answerer is willing to receive. For streams
marked as sendonly in the answer, the "m=" line MUST contain at least
one media format the answerer is willing to send from amongst those
listed in the offer. For streams marked as sendrecv in the answer,
the "m=" line MUST contain at least one codec the answerer is willing
to both send and receive, from amongst those listed in the offer.
The stream MAY indicate additional media formats, not listed in the
corresponding stream in the offer, that the answerer is willing to
send or receive (of course, it will not be able to send them at this
time, since it was not listed in the offer). "
Is this how it is usually detected and what every SIP client does?
Thanks,
José
_______________________________________________
Sipping mailing list https://www1.ietf.org/mailman/listinfo/sipping
This list is for NEW development of the application of SIP
Use sip-implementors at cs.columbia.edu for questions on current sip
Use sip at ietf.org for new developments of core SIP