Re: [Sip-implementors] renegotiate rtp <--> srtp in mid call
J C Sunil Kumar Reddy writes: > Can we renegotiate a rtp call to srtp and vice-versa in midcall? May be > with re-INVITE or UPDATE message? Is it a valid scenario if srtp is not > mandated? > > In my scenario, server and client both supports RTP and SRTP and initially > call is negotiated to srtp call. Midway, server is sending a re-INVITE > asking to change to RTP. What should client do in this case? Accept, as > SRTP is not mandatory or drop the call? The most relevant RFC is 3264, section 8, "Modifying the Session". It is permitted for either endpoint to request to modify the session. Whether it does this by re-INVITE or UPDATE does not matter. As 3264 says, almost any aspect of a media description (an "m-line section" of the SDP) can be changed. My memory is that the only part that cannot be changed is the media type, e.g., "m=audio", "m=video". In particular, the "transport" (RTP/AVP, RTP/SAVP) can be changed. The difficulty is that the other endpoint can always reject the change for any reason it wants. So although it is valid to request the change, there is no assurance that the change will be accepted. In general, you should avoid dropping the call if the remote end requests something that that the UA does not want to act on. There is almost always a straightforward way to reject the request and the requestor should handle this correctly. Dropping the call should be done only if there is evidence that the two UAs are no longer able to continue communicating for some reason. Dale ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
Re: [Sip-implementors] renegotiate rtp <--> srtp in mid call
On 9/10/19 9:14 AM, J C Sunil Kumar Reddy wrote: Hi All, Can we renegotiate a rtp call to srtp and vice-versa in midcall? May be with re-INVITE or UPDATE message? Is it a valid scenario if srtp is not mandated? In my scenario, server and client both supports RTP and SRTP and initially call is negotiated to srtp call. Midway, server is sending a re-INVITE asking to change to RTP. What should client do in this case? Accept, as SRTP is not mandatory or drop the call? Which RFC should I refer for this scenario and what does it say in this case? I don't know of any document that discusses this exact scenario. But this is simply a particular case of replacing one medium with another, and so it entirely appropriate. While you should be able to just change the existing m= line, you might want to consider just disabling that one with port=0 and adding a different m= line for the srtp. Or you could use one O/A exchange to disable the old media with port=0 and then use yet another O/A to re-establish it with srtp and a valid port. But "just changing it" ought to work. Thanks, Paul ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
[Sip-implementors] renegotiate rtp <--> srtp in mid call
Hi All, Can we renegotiate a rtp call to srtp and vice-versa in midcall? May be with re-INVITE or UPDATE message? Is it a valid scenario if srtp is not mandated? In my scenario, server and client both supports RTP and SRTP and initially call is negotiated to srtp call. Midway, server is sending a re-INVITE asking to change to RTP. What should client do in this case? Accept, as SRTP is not mandatory or drop the call? Which RFC should I refer for this scenario and what does it say in this case? Regards, Sunil ___ Sip-implementors mailing list Sip-implementors@lists.cs.columbia.edu https://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
