Hi, Thanks for the response. I have a basic question, considering 3 views.
1. Suppose terminating endpoint does not support 183, and sends the early media for IVR, should UAC, play the media treated as if early media. 2. Any case the terminating endpoint must support 183 and send the media after it sends. The 100rel is optional, in this case the sequence of 183 and early media is not guaranteed. Should UAC prepare media irrespective of call progress message? 3. In case of 100Rel calling UA requires the option 100rel, should terminating endpoint MUST send always after PRACK is received? Specifically, if we want deploy a firewall between originating and terminating endpoints, should we allow early media after INVITE without 183 is being seen? Thanks, Anil -----Original Message----- From: Dean Willis [mailto:[EMAIL PROTECTED] Sent: Tuesday, December 07, 2004 2:05 PM To: Anil Bollineni Cc: Syed, Mohammad Rafi; Fatih "Ey|p" NAR; [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Sip] Doubts in SIP/SDP On Dec 6, 2004, at 4:58 PM, Anil Bollineni wrote: > Hi, > Thanks for the response. Actually I may confuse you. Specifically, > if the callee sends the media after 183, and 183 is lost, will the > caller should accept the media, because it don't know codec should the > media expect or caller should decode whatever is coming, by finding > the codecs in the list it offers. Sorry I don't know that it is the > basic requirement, but in RFC can I find the statements that tell the > media starts after the SDP negotiation is completed. > I believe the rule is "the caller MUST be ready to receive any media described in the SDP of an INVITE immediately after sending that INVITE." This is how we get all those neat early-media use-cases. Note that this use case frequently occurs when placing a call from SIP through a PSTN gateway (with ISDN) that terminates on a PSTN IVR, like that used by American Airlines. The "answer" SDP arrives in a 200OK that is sent by the gateway after the PSTN circuit moves into connected state. However, the phone circuit is not in the "connected" state for the first round of engagement with the IVR. The gateway in my office will only tolerate this for about 32 seconds before it gives up on the INVITE, so I have to navigate the first level of the IVR quickly. -- Dean _______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
