One document matched: draft-ohta-notasip-01.txt
Differences from draft-ohta-notasip-00.txt
INTERNET DRAFT Masataka Ohta
draft-ohta-notasip-01.txt Tokyo Institute of Technology
Kenji Fujikawa
Kyoto University
10 April 1998
Nothing Other Than a Simple Internet Phone (NOTASIP)
Status of this Memo
This document is an Internet-Draft. Internet-Drafts are working
documents of the Internet Engineering Task Force (IETF), its areas,
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Abstract
This memo describes a simple protocol for Internet phone without QoS
guarantee.
1. The architectural Principle of the Internet Phone
Fortunately enough, Internet is the install base of data
communication that other network technologies must find someway to
interoperate with the Internet to survive a little longer.
However, in the world of phone communication today, POTS is the
install base. For the Internet to replace POTS within a few years, it
is important that Internet phone interoperates with POTS.
So, the primary requirement to the Internet phone at this early stage
is that it should be able to interoperate with a dumb analog phone,
which constitutes the install base.
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Historically, telephone companies in different nations tried hard to
make their system not to interoperate smoothly to protect their
market. None the less, or as a result, protocols to interoperate POTS
is well developed. The protocols must be constructed over voice, the
only common transport over different phone systems. A notable
protocol is operator assisted call. However, as human intervention
costs a lot, most POTS support tone dialing capability for the least
digital communication capability.
Note that complex capabilities of digital phones, which is
disappearing, have nothing to do with the install base and are
ignored in this memo.
Possible complex capabilities of Internet phone such as multiparty
teleconferencing, which is hard to operate over voice, are also
ignored in this memo. It should be noted that there is multiparty
teleconferencing service already available through POTS, simulation
of which over the Internet phone is not difficult without complex
Internet protocol.
POTS is the install base worth considering. As it is difficult for
human being to generate or recognize IP packets over POTS with voice
or dial tones, protocols needs complex exchange of IP packets should
be considered seriously only after we don't have to interoperate with
POTS,
It is assumed that the operating system support a notion of connected
UDP socket [UNIX].
2. Caller Initiate the Call
The caller host somehow (through SDP URL [SDP, SDPURL] of callee's
home page, for example) finds the callee's IP address, UDP port
number (with default port number of <to be assigned by IANA>) and
desired encoding.
The caller host opens a UDP socket and start sending properly encoded
UDP packets of voice.
3. Callee Accept the Call
The callee host receiving a UDP packet from someone opens a new
connected UDP socket to the callers UDP port using a new source port
number and the same IP address as the destination address of caller's
packet.
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The callee host, then, start ringing the phone to notify the
existence of a call to the callee person. The ringing tone should
also be sent to the caller.
If the callee host do not want to accept simultaneous call, it may
suspend the UDP port used to accept the call. Then, if the port is
already connected to someone else, ICMP error packet is returned,
which makes the caller host generate a busy signal to the caller
person.
4. Connection Established
If the callee person holds up a headset, the callee host should send
the voice of the callee person to the caller host. The caller host
receives a first packet and, confirming the source IP address of the
packet is callee host's, connects its UDP port to the callee's
sending port and the call is established.,
5. Call Termination
To terminate the call, the caller or callee host close the socket.
The same port number should not be used again until 256 (maximum IPv4
TTL) + 30 seconds passes.
6. Interoperation with PSTN
Interoperation with PSTN is performed over voice, the only common
transport, with operator assistance, dial tone or anything. The
exact protocol over the voice is service provider dependent and MUST
NOT be standardized. IETF MUST NOT define a standard on natural
language messages to/from telephone operators nor calling card
syntax.
7. Error Conditions
If the connected UDP socket can not be created or the socket
generates some error, the call terminates.
If the caller host receives a UDP packet from someone other than the
callee host before the call established, they should be ignored.
If there is no packets received to a port for 30 seconds, the call
terminates.
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8. References
[SDP] Mark Handley and Van Jacobson, "SDP: Session Description Proto-
col," Internet Draft draft-ietf-mmusic-sdp-05.txt (work in
progress), Nov 1997.
[SDPURL] (to appear)
[UNIX] See UNIX manuals.
9. Security Considerations
The security of POTS accounting is often based on 4 digit password or
plain credit number and is quite poor. Moreover, it is, in general,
impossible to know the phone number of the caller. But these are the
accepted security of the phone system.
Best effort Internet phone is basically free (except for a flat rate
portion) that no serious security consideration is necessary as a
phone system. A possible denial of service attack can be based on
forged caller source IP address but is a lot more harmless than the
similar attack with POTS.
Authors' Addresses
Masataka Ohta
Computer Center
Tokyo Institute of Technology
2-12-1, O-okayama, Meguro-ku, Tokyo 152, JAPAN
Phone: +81-3-5734-3299
Fax: +81-3-5734-3415
EMail: mohta@necom830.hpcl.titech.ac.jp
Kenji Fujikawa
Faculty of Information Science
Kyoto University
Yoshidahonmachi, Sakyo Ku, Kyoto City, 606-01, JAPAN
Phone: +81-75-753-5387
Fax: +81-75-751-0482
EMail: magician@kuis.kyoto-u.ac.jp
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