draft-ietf-pint-pre-implement-01.txt   draft-ietf-pint-pre-implement-02.txt 
skipping to change at page 1, line 19 skipping to change at page 1, line 19
F. Burg F. Burg
A. DeSimone A. DeSimone
K. T. Tewani K. T. Tewani
AT&T Labs AT&T Labs
P. Davidson P. Davidson
Nortel Nortel
H. Schulzrinne H. Schulzrinne
Columbia University Columbia University
K. Vishwanathan K. Vishwanathan
Isochrome Isochrome
Expires in Six Months August 1998
Toward the PSTN/Internet Inter-Networking Toward the PSTN/Internet Inter-Networking
--Pre-PINT Implementations --Pre-PINT Implementations
<draft-IETF-pint-pre-implement-01.txt> <draft-IETF-pint-pre-implement-02.txt>
Status of this Memo Status of this Memo
This document is an Internet-Draft. Internet-Drafts are working This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas, documents of the Internet Engineering Task Force (IETF), its areas,
and its working groups. Note that other groups may also distribute and its working groups. Note that other groups may also distribute
working documents as Internet-Drafts. working documents as Internet-Drafts.
Internet-Drafts are draft documents valid for a maximum of six months Internet-Drafts are draft documents valid for a maximum of six months
and may be updated, replaced, or obsoleted by other documents at any and may be updated, replaced, or obsoleted by other documents at any
skipping to change at page 40, line 26 skipping to change at page 40, line 26
Extensions of SIP to allow third-party signaling (e.g., for click- Extensions of SIP to allow third-party signaling (e.g., for click-
to-dial-back services, fully meshed conferences and connections to to-dial-back services, fully meshed conferences and connections to
Multipoint Control Units (MCUs), as well as mixed modes and the Multipoint Control Units (MCUs), as well as mixed modes and the
transition between those) have been specified. transition between those) have been specified.
SIP addresses users by an email-like address and re-uses some of the SIP addresses users by an email-like address and re-uses some of the
infrastructure of electronic mail delivery such as DNS MX records or infrastructure of electronic mail delivery such as DNS MX records or
using SMTP EXPN for address expansion. SIP addresses (URLs) can also using SMTP EXPN for address expansion. SIP addresses (URLs) can also
be embedded in Web pages. SIP is addressing-neutral, with addresses be embedded in Web pages. SIP is addressing-neutral, with addresses
expressed as URLs of various types such as SIP, H.323 or telephone expressed as URLs of various types such as SIP, H.323 or telephone
(E.164). An example of a telephone URL might be (E.164). A purely representational example of a SIP URL might be
sip://12125551212@foo.example.com, where foo.example.com is the host sip:12125551212@foo.example.com, where foo.example.com is the host
serving as a gateway into the PSTN. serving as a gateway into the PSTN.
SIP is independent of the packet layer and only requires an SIP is independent of the packet layer and only requires an
unreliable datagram service, as it provides its own reliability unreliable datagram service, as it provides its own reliability
mechanism. While SIP typically is used over UDP or TCP, it could, mechanism. While SIP typically is used over UDP or TCP, it could,
without technical changes, be run over IPX, or carrier pigeons, ATM without technical changes, be run over IPX, or carrier pigeons, ATM
AAL5 or X.25, in rough order of desirability. AAL5 or X.25, in rough order of desirability.
SIP can set up calls "out-of-band". For example, while the SIP SIP can set up calls "out-of-band". For example, while the SIP
protocol exchanges use IP, plus UDP or TCP, the actual data transport protocol exchanges use IP, plus UDP or TCP, the actual data transport
skipping to change at page 42, line 17 skipping to change at page 42, line 17
made. made.
A SIP client needs to convey two addresses to the PSTN gateway: the A SIP client needs to convey two addresses to the PSTN gateway: the
party making the call and the party to be called. (The party to be party making the call and the party to be called. (The party to be
billed also needs to be identified; this can either be done by a SIP billed also needs to be identified; this can either be done by a SIP
header or by having the server look up the appropriate party based on header or by having the server look up the appropriate party based on
the two parties. This aspect is for further study.) the two parties. This aspect is for further study.)
Described below are three ways these addresses can be conveyed in Described below are three ways these addresses can be conveyed in
SIP. In the example, the address of party A is +1-212-555-1234 and SIP. In the example, the address of party A is +1-212-555-1234 and
that of party B is +1-415-555-1200. that of party B is +1-415-555-1200. (The URL types in this and other
examples are representational; they may but do not have to exist.)
(1) The two PSTN addresses are contained in the To header (and (1) The two PSTN addresses are contained in the To header (and
request-URI) and an Also header. For example: request-URI) and an Also header. For example:
INVITE sip://1-212-555-1234@pbx.example.com INVITE sip:1-212-555-1234@pbx.example.com
To: phone://1-212-555-1234 To: phone:1-212-555-1234
From: j.doe@example.com From: sip:j.doe@example.com
Content-type: application/sdp Content-type: application/sdp
Call-ID: 19970721T135107.25.181@foo.bar.com Call-ID: 19970721T135107.25.181@foo.bar.com
Also: phone://+1-415-555-1200 Also: phone:+1-415-555-1200
v=0 v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5 o=user1 53655765 2353687637 IN IP4 128.3.4.5
c=PSTN E.164 +1-415-555-1200 c=PSTN E.164 +1-415-555-1200
t=0 0 t=0 0
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
In that case, the gateway first connects to party A and then party B, In that case, the gateway first connects to party A and then party B,
but without waiting for A to accept the call before calling B. but without waiting for A to accept the call before calling B.
(2) Parties A and B are indicated by separate invitations. This (2) Parties A and B are indicated by separate invitations. This
allows the gateway to make sure that party A is indeed available allows the gateway to make sure that party A is indeed available
before calling party B. After calling party A, the gateway could before calling party B. After calling party A, the gateway could
play an announcement indicating that the call is being connected play an announcement indicating that the call is being connected
using, for example, RTSP with appropriate Conference header using, for example, RTSP with appropriate Conference header
indicating the call. indicating the call.
INVITE sip://1-212-555-1234@pbx.example.com INVITE sip:1-212-555-1234@pbx.example.com
To: phone://1-212-555-1234 To: phone:1-212-555-1234
From: j.doe@example.com From: sip:j.doe@example.com
Content-type: application/sdp Content-type: application/sdp
Call-ID: 19970721T135107.25.181@foo.bar.com Call-ID: 19970721T135107.25.181@foo.bar.com
... ...
INVITE sip:1-415-555-1200@pbx.example.com
INVITE sip://1-415-555-1200@pbx.example.com To: phone:+1-415-555-1200
To: phone://+1-415-555-1200 From: sip:j.doe@example.com
From: j.doe@example.com
Content-type: application/sdp Content-type: application/sdp
Call-ID: 19970721T135107.25.181@foo.bar.com Call-ID: 19970721T135107.25.181@foo.bar.com
... ...
(3) The two PSTN addresses are conveyed in the To header of the SIP (3) The two PSTN addresses are conveyed in the To header of the SIP
request and the address in the SDP media description. Thus, a request request and the address in the SDP media description. Thus, a request
may look as follows: may look as follows:
INVITE sip://1-212-555-1234@pbx.example.com INVITE sip:1-212-555-1234@pbx.example.com
To: phone://1-212-555-1234 To: phone:1-212-555-1234
From: j.doe@example.com From: sip:j.doe@example.com
Content-type: application/sdp Content-type: application/sdp
Call-ID: 19970721T135107.25.181@foo.bar.com Call-ID: 19970721T135107.25.181@foo.bar.com
v=0 v=0
o=user1 53655765 2353687637 IN IP4 128.3.4.5 o=user1 53655765 2353687637 IN IP4 128.3.4.5
c=PSTN E.164 +1-415-555-1200 c=PSTN E.164 +1-415-555-1200
t=0 0 t=0 0
m=audio 0 RTP/AVP 0 m=audio 0 RTP/AVP 0
Here, pbx.example.com is the name of the PSTN gateway; the call will Here, pbx.example.com is the name of the PSTN gateway; the call will
 End of changes. 10 change blocks. 
19 lines changed or deleted 21 lines changed or added

This html diff was produced by rfcdiff 1.34. The latest version is available from http://tools.ietf.org/tools/rfcdiff/