[EMAIL PROTECTED] wrote:
I have some issues regarding the "TEL" uri implementation in SIP.
There do seem to be a variety of opinions on this. Here are mine.
1. What shoud be the valid TEL uri
Well, that depends on how aggressive you want to be with the standards.
If you only do RFCs, then you want to conform to RFC 2806. But that is pretty long in the tooth, and has a number of problems.
If you want to be on the bleeding edge, you should be conforming to draft-ietf-iptel-rfc2806bis-09.
2.What shoud be the request header and "to" and "from" field in INVITE message
All of those MAY contain tel: URIs.
The most important question is probably how a UAC SHOULD initialize a request when it has a phone number it wants to call, and/or it wants to be known by a phone number.
Consider the destination first:
Suppose the UAC knows the complete global number that is to be called. Then IMO that SHOULD be placed in the To header, as a global tel: uri. What then goes into the request uri is widely argued:
- you could put the tel: uri there too. Then deliver the request to some proxy for further evaluation. (That is my recommendation if you normally have an outgoing proxy doing the heavy lifting for you.)
- you could use ENUM to translate the number. If successful, put one of the results from that in the request URI. (This is probably your best bet if you have no proxy capable of helping you.)
- you could transform the tel: uri to a sip: uri with the same string in the user part, some domain you have reason to believe can deal with this, and a user=phone parameter. (I find little/no benefit from this.)
If what you have is a dial string rather that a full global phone number, and you want somebody else to figure out what it means, then the situation becomes more complicated. I won't go on unless you really want to open that can of worms.
Paul
2.What are the requirement to complete call flow for a tel uri
No different than any other call. If the callee is directly accessible via sip then the request uri will probably be translated one or more times (into sip uris) before reaching the destination.
If the callee is out on the pstn, then the call will end up landing on a gateway that will launch it into the pstn. The request uri could still be a tel: uri when it hits the gw, or it might have been transformed into a sip uri first.
Paul
_______________________________________________ Sip-implementors mailing list [EMAIL PROTECTED] http://lists.cs.columbia.edu/mailman/listinfo/sip-implementors
