One document matched: draft-rosenberg-sip-3pcc-01.txt
Differences from draft-rosenberg-sip-3pcc-00.txt
Internet Engineering Task Force SIP WG
Internet Draft Rosenberg/Peterson/Schulzrinne/Camarillo
draft-rosenberg-sip-3pcc-01.txt dynamicsoft,Level3,Columbia U.,Ericsson
November 22, 2000
Expires: May 2001
Third Party Call Control in SIP
STATUS OF THIS MEMO
This document is an Internet-Draft and is in full conformance with
all provisions of Section 10 of RFC2026.
Internet-Drafts are working documents of the Internet Engineering
Task Force (IETF), its areas, and its working groups. Note that
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The list of current Internet-Drafts can be accessed at
http://www.ietf.org/ietf/1id-abstracts.txt
The list of Internet-Draft Shadow Directories can be accessed at
http://www.ietf.org/shadow.html.
Abstract
This document discusses the usage of the Session Initiation Protocol
(SIP) for third party call control. Third party call control refers
to the ability of one entity to create a call in which communications
is actually between other parties. We present a SIP mechanism for
accomplishing third party call control that does not require any
extensions or changes to SIP.
1 Introduction
In the traditional telephony context, third party call control allows
one entity (which we call the controller) to set up and manage a
communications relationship between two or more other parties. Third
party call control is often used for operator services (where an
operator creates a call that connects two participants together), and
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conferencing.
On the Internet, a wider range of services are enabled through a
third party session control mechanism. This is because other IP
applications, such as web, email, presence, instant messaging, and
chat can now be brought into the picture. An excellent example is
click-to-dial. This service allows a user to click on a web page when
they wish to speak to a customer service representative. The web
server then creates a call between the user and a customer service
representative. The call can be between two phones, a phone and an IP
host, or two IP hosts.
In order to support third party call control applications, a
mechanism is needed that allows a controller to create, modify, and
terminate calls with other entities. In this document, we present a
mechanism using the Session Initiation Protocol (SIP) [1] which
allows a controller to execute third party services. The mechanism is
not an extension to SIP. It is merely an application of the tools
enabled through RFC 2543. A controller can create calls between any
entity that contains a normal SIP user agent. After desribing the
mechanism, we present three third party services which take
advtantage of this mechanism. One is click-to-dial, the second is a
feature that enables a mid-call announcement for credit card
authorization , and the third is a timed conference bridge
initiation.
2 Third Party Control
The basic idea behind the third party mechanism is simple. Consider
first the case of just connecting two users in a call. The controller
first sends an INVITE to the first user whose phone is to ring. This
is a standard INVITE, but its SDP contains a single audio media line,
with one codec, a random port number (but not zero), and a connection
address of 0.0.0.0. This creates an initial media stream "on hold".
When the first user answers, the controller sends an ACK. It then
generates a second INVITE. This INVITE is addressed to the second
user to be connected in the call. This INVITE contains the SDP as
received from the 200 OK of the first user. When the 200 OK to this
second INVITE arrives, the controller ACK s it, takes the SDP, and
then re-INVITEs the first user with this updated SDP. A flow diagram
for this mechanism is given in 1.
At this point, both participants believe they are in a single point-
to-point call with some control system (assuming the controller
identified itself as such in the From field of the INVITE). However,
they are exchanging media directly with each other, rather than with
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A Controller B
| INV held SDP | |
|<------------------| |
| | |
| 200 SDP A | |
|-----------------> | INV SDP A |
| ACK |----------------->|
|<----------------- | |
| | 200 SDP B |
| |<-----------------|
| | |
| | ACK |
| INV SDP B |----------------->|
|<------------------| |
| 200 OK SDP A | |
|------------------>| |
| ACK | |
|<------------------| |
| | RTP |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
Figure 1: Basic Third Party Call Control
the controller. The result is that the controller has set up a call
between both participants.
Since the controller is still a central point for signaling, it now
has complete control over the call. If it receives a BYE from one of
the participants, it can create a new BYE and hang up with the other
participant. This is shown in 2.
As an alternative, when the controller receives a BYE from A, it can
generate a new INVITE to a third party, C, using the SDP from B. When
the 200 OK arrives from C, the controller sends a re-INVITE to B,
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A Controller B
| | |
| | |
| BYE From A | |
|-----------------> | BYE From Cont. |
| 200 OK |----------------> |
|<----------------- | 200 OK |
| |<---------------- |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
| | |
Figure 2: Hanging Up with 3PCC
using the SDP from C. If the 200 OK to the re-INVITE contains the
same SDP as it used in the INVITE to C, the controller has
sucessfully connected B to C, transparently to B. A call flow for
this is shown in 3.
From here, new parties can be added, removed, transferred, and so on,
as the controller sees fit.
The general idea behind the mechanism is that there is a point to
point SIP relationship between each participant and the controller.
However, by passing the SDP it receives from one participant to
another, it can causes users to actually communicate with each other
rather than the controller.
3 Third party call control and SDP preconditions
In unicast sessions there is a number of media streams flowing
between two entities. In order to perform resource reservation it is
necessary to know the session descriptions from both parties. When
third party call control is performed the information needed to
establish the QoS required is not available from the beginning. The
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A Controller B C
| | | |
| | | |
| BYE From A | | |
|-----------------> | INV SDP B | |
| 200 OK |------------------------------------>|
|<----------------- | | 200 SDP C |
| |<------------------------------------|
| | ACK | |
| |------------------------------------>|
| | INV SDP C | |
| |----------------->| |
| | 200 SDP B | |
| |<-----------------| |
| | ACK | |
| |----------------->| |
| | | |
| | | RTP |
| | | xxxxxxxxxxxxxxxx |
| | | xxxxxxxxxxxxxxxx |
| | | |
| | | |
| | | |
| | | |
Figure 3: Alternative to Hangup
call flow shown in Figure 4 shows how the exchange of SDPs between
both parties can be performed.
The controller INVITEs A in (1). At this point of time there is no
information available about codecs to be used port numbers or IP
addresses. The SDP of this INVITE just contains SDP preconditions and
the media stream types (audio, video, etc...). As specified in [2],
the called UAS returns a 183 immediately containing SDP information
needed for QoS signaling (2).
INVITE (3) contains the SDP received from A. This INVITE is sent to
B. When B responses with (4) 183 it is ready to perform resource
reservation. However, B will not start resource reservation until the
PRACK (7) is received. This allows B's SDP to be sent to A in (5).
This way both parties have all the information needed to perform
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resource reservation. Note that, since reliable provisional responses
are used [3], the 183 (2) is retransmitted until the PRACK (5)
arrives from the controller. This PRACK is transmitted only when the
183 arrives from B (4). Fortunately, this 183 is generated
automatically, so that the first 183 (2) should not be retransmitted
that much, if at all.
The PRACK matching (2) is sent at (5). This PRACK is not sent before
because it is used to send B's SDP to A. The controller does not get
this information until (4).
When the preconditions from B to the controller and from A to the
controller are met two COMETs are received (9) and (11). At this
point of time is up to the controller to let the session
establishment go on sending a COMET to A (13). When A accepts joining
the session (15), a COMET (16) is sent to B so B is alerted.
4 Click to Dial
The first application of this capability we discuss is click to dial.
In this service, a user is browsing the web page of an e-commerce
site, and would like to speak to a customer service representative.
They click on a link, and the phone on the desk (a normal telephone)
rings. When the user picks up, the phone of the customer service
representative (an IP phone) rings. When they pick up, the service
representative is talking to the user.
We assume for purposes of this discussion that the web server is
actually an applications server that contains an http interface. In
this case, when the user clicks on the URL, the application server
knows, through cookies or some other state mechanism, the addresses
of the participants to be connected.
The call flow for this service is given in 5. Note that it is
identical to that of Figure 1, with the exception that the service is
triggered through an http GET request when the user clicks on the
link.
We note that this service can be provided through other mechanisms,
namely PINT [4]. However, there are numerous differences between the
way in which the service is provided by pint, and the way in which it
is provided here:
o The pint solution enables calls only between two PSTN
endpoints. The solution described here allows calls between
PSTN phones (through SIP enabled gateways) and native IP
phones.
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Controller A B
| (1) INVITE | |
|------------------>| |
| (2) 183 SDP A | |
|<------------------| |
| (3) INVITE SDP A | |
|------------------------------------->|
| (4) 183 SDP B | |
|<-------------------------------------|
| (5) PRACK SDP B | |
|------------------>| |
| (6) 200 OK (PRACK)| |
|<------------------| |
| (7) PRACK | |
|------------------------------------->|
| (8) 200 OK (PRACK)| |
|<-------------------------------------|
| (9) COMET | |
|<-------------------------------------|
|(10) 200 OK (COMET)| |
|------------------------------------->|
| (11) COMET | |
|<------------------| |
|(12) 200 OK (COMET)| |
|------------------>| |
| (13) COMET | |
|------------------>| |
|(14) 200 OK (COMET)| |
|<------------------| |
|(15) 200 OK (INVITE) |
|<------------------| |
| (16) COMET | |
|------------------------------------->|
|(17) 200 OK (COMET)| |
|<-------------------------------------|
|(18) 200 OK (INVITE) |
|<-------------------------------------|
| (19) ACK | |
|------------------>| |
| (20) ACK | |
|------------------------------------->|
| | |
Controller A B
Figure 4: Call Flow for Preconditions
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PSTN Controller Customer Users PC
GW Service
Representative
| | HTTP GET | |
| |<-----------------+-------------------|
| | 200 OK | |
| |------------------+------------------>|
| | | |
| | | |
| | | |
| | | |
| INV SDP held | | |
|<------------------| | |
| | | |
| 200 SDP A | | |
|-----------------> | INV SDP A | |
| |----------------->| |
| | | |
| | 200 SDP B | |
| |<-----------------| |
| | | |
| | ACK | |
| INV SDP B |----------------->| |
|<------------------| | |
| 200 SDP A | | |
|------------------>| | |
| ACK | | |
|<------------------| | |
| | | |
| | RTP | |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx | |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
| | | |
Figure 5: Click to Dial Call Flow
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o When used for calls between two PSTN phones, the solution here
may result in a portion of the call being routed over the
Internet. In pint, the call is always routed only over the
PSTN. This may result in better quality calls with the pint
solution, depending on the codec in use and QoS capabilities
of the network routing the Internet portion of the call.
o The PINT solution requires extensions to SIP (PINT is an
extension to SIP), whereas the solution described here is done
with baseline SIP.
o The PINT solution allows the controller (acting as a PINT
client) to "step out" once the call is established. The
solution described here requires the controller to maintain
call state for the entire duration of the call.
5 Mid-Call Announcement Capability
The third party call control mechanism described here can also be
used to enable mid-call announcements. The call is set up by the
controller as desribed above.
It is actually not necessary for the controller to set up
the call. However, if a participant initiates the call, the
controller must step in as a virtual UAC/UAS, and act as a
termination and re-initiation point
Perhaps the call is through a payphone, in which case the controller
determines that the call is to be terminated after some amount of
time if the user doesn't add more money to the phone. When this timer
expires, the controller places the called party on hold. It then
sends an INVITE to the media server which will be collecting digits.
It then sends a re-INVITE to the user on the payphone, connecting its
media streams with the media server. The media server plays an
announcement, and prompts the user to enter a credit card number, for
example. After collecting the number and validating the card, if the
call can continue, the media server hangs up. The controller takes
this as a cue and reconnects the user to the original called party,
and takes the original called party off hold.
A call flow for this service is shown in 6.
We have assumed that the media server and the controller have agreed,
ahead of time, that a hangup implies that the desired service
(extending the lifetime of the call) has succeeded. This is
effectively allowing a call control interface between the controller
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and the media server. Parameters needed between the elements, such as
the new expiration of the call, can be passed in the BYE. A separate
draft, forthcoming, will discuss call control interfaces to media
services in more detail.
6 Timed Conference Intitation
In this service, a conference bridge is booked for some number of
participants. In order to make sure the conference begins on time,
the conference bridge will call each participant at the time of the
call. If a participant doesn't answer, the bridge tries to contact
them again (unless they call in) five minutes later.
In the call flow described here, we assume that the controller acts
as the media bridge. This is not strictly necessary; some kind of
control interface could be used to separate the media function from
the controller.
The call flow, shown in 7, is, not surprisingly, remarkably like that
of Figure 1. The only difference is that the SDP listed in the INVITE
s generated by the controller always contain SDP that points to the
conference bridge, rather than one of the other participants. In the
call flow diagram, user 1 is invited first, then user 2, and then
user 3. User 3 is not available, but is called again five minutes
later.
7 Implementation Notes
Most of the work involved in supporting third party call control is
within the controller. A standard SIP UA should be controllable in
the mechanism described here. However, the mechanism relies on a few
features that might not be implemented. As such, we strongly
recommend implementors of user agents to support the following:
o Re-invites that change the port to which media should be send
o Re-invites that change the connection address
o Re-invites that add a media stream
o Re-invites that remove a media stream (setting its port to
zero)
o Re-invites that add a codec amongst the set in a media stream
o Hold (connection address of zero)
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Payphone Controller Called Media
"A" Party Server
"B" "C"
| RTP | | |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx| |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx| |
| | INV SDP 0 (hold) | |
| |----------------->| |
| | 200 OK | |
| |<-----------------| |
| | ACK | |
| |----------------->| |
| | INV SDP A | |
| |------------------------------------->|
| | | 200 SDP C |
| INV SDP C |<-------------------------------------|
|<------------------| ACK | |
| 200 SDP A |------------------------------------->|
|------------------>| | |
| ACK | | |
|<------------------| | |
| | | |
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx|
|xxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxxx|
| | | BYE |
| |<-------------------------------------|
| | 200 OK | |
| |------------------------------------->|
| | INV SDP A | |
| |----------------->| |
| | 200 SDP B | |
| INV SDP B |<-----------------| |
|<------------------| ACK | |
| 200 SDP A |----------------->| |
|------------------>| | |
| ACK | | |
|<------------------| | |
| | | |
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User 1 Controller User 2 User 3
"A" "X" "B" "C"
| INV SDP X | | |
|<------------------| | |
| | INV SDP X | |
| 200 SDP A |----------------->| |
|-----------------> | | |
| | 200 SDP B | |
| ACK |<-----------------| |
|<------------------| | |
| | ACK | |
| |----------------->| |
| | | |
| | INV SDP X | |
| |--------------------------------------->|
| | | 408 Timeout |
| |<---------------------------------------|
| | ACK | |
| |--------------------------------------->|
| | | |
| | | |
| | INV SDP X | |
| |--------------------------------------->|
| | | 408 Timeout |
| |<---------------------------------------|
| | ACK | |
| |--------------------------------------->|
| | | |
| | | |
| | | |
| | | |
| | | |
Figure 7: Timed Conference Initiation Call Flow
o Initial invites on hold
In addition, note that in Figure 1, the controller sends a re-INVITE
to A with the SDP from B. The response to this re-INVITE is a 200 OK
that contains "SDP A". If the SDP returned in the re-INVITE response
were not the same, the controller would need to initiate a re-INVITE
to B with that new SDP. This means that a UA which returns a
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different SDP (for example, by changing the ports) in the response to
every re-INVITE will trigger an infinite re-INVITE loop from the
controller to each controller entity. As such, it is STRONGLY
RECOMMENDED that if a re-INVITE does not require a UAS to modify the
formulation of the SDP description for a specific stream in the
response, the SDP description of that stream in the response MUST be
the same. In other words, if a UA was in a session with a single
stream, using codec A, and it received a re-INVITE modifying the port
to send to, the re-INVITE response MUST be the same as it was to the
initial request. However, if the re-INVITE forces the UAS to change
codecs, it is acceptable in that case to use a different port in the
SDP in the response.
In a previous draft, the flow of Figure 1 used a delayed ACK instead
of a re-INVITE to update the SDP with the first user. The delayed ACK
approach results in fewer messages (a savings of two), but has
timeout problems. Specifically, If the second user does not answer
within 32 seconds, the first user will timeout (as it did not receive
an ACK), and the call will not be established. Since this will happen
in practice, we strongly recommend the procedure described above
rather than the delayed ACK mechanism.
8 Security Considerations
The mechanism described here introduces several security
considerations. The first issue is the calling party identities
delivered to the participants which the controller invites. The
controller could indicate that the call is from itself (From:
sip:controller@company.com), but in many cases, the service is more
usable if it "spoofs" the identity of the participant that is
actually calling. However, to differentiate legitimate use of 3pcc
from real attacks, user agents SHOULD authenticate the requests. The
controller MUST sign the request as itself, not as A or B. This will
allow both parties to know that the call is actually being
established through a controller. User agents SHOULD be configured to
authorize requests from entities known to be controllers.
Note that this will result in SIP messages whose From field does not
match the identity of the signator (as indicated in the signed-by
field of the request).
The third party mechanism can also have an impact on encryption of
the media that is part of the session. If negotiation of session keys
is done through some kind of key exchange within SIP, the controller
will, in all likelihood, not be able to set it up so that
participants in the call arrive at the same key. This means that the
controller may need to act as an RTP translator, decrypting with one
key and re-encrypting with another.
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Third party call control has unfortunate interactions with NATs and
firewalls. The problems arise when the controller is on one side of a
firewall/NAT that is being controlled by a proxy [5] [6] that
receives the controller's requests, and the controlled users are on
the other side. Pinholes in the firewall may be opened when, in fact,
the media does not pass through the firewall. One way to avoid this
is for the firewall controlling proxy to recognize that the address
of the media is not within its private network, and so not perform
NAT or firewall control in those cases.
9 Conclusions
We have presented a basic third party call control mechanism that
uses SIP. This mechanism does not require any extensions to SIP and
is completely backwards compatible.
10 Changes since -00
o Modified basic flow to use re-INVITE instead of delayed ACK.
o Included preconditions interactions.
o Added implementation considerations section.
o Updated security considerations to talk about NAT/firewall
control, and to introduce the notion of signing requests when
the signator is not the user in the From field.
11 Authors Addresses
Jonathan Rosenberg
dynamicsoft
72 Eagle Rock Avenue
First Floor
East Hanover, NJ 07936
email: jdrosen@dynamicsoft.com
Jon Peterson
Level 3 Communications
1025 Eldorado Blvd
Broomfield, CO 80021
email: Jon.Peterson@level3.com
Henning Schulzrinne
Columbia University
M/S 0401
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1214 Amsterdam Ave.
New York, NY 10027-7003
email: schulzrinne@cs.columbia.edu
Gonzalo Camarillo
Ericsson
Advanced Signalling Research Lab.
FIN-02420 Jorvas
Finland
Phone: +358 9 299 3371
Fax: +358 9 299 3052
Email: Gonzalo.Camarillo@ericsson.com
12 Bibliography
[1] M. Handley, H. Schulzrinne, E. Schooler, and J. Rosenberg, "SIP:
session initiation protocol," Request for Comments 2543, Internet
Engineering Task Force, Mar. 1999.
[2] B. Marshall et al. , "Integration of resource management and
SIP," Internet Draft, Internet Engineering Task Force, Nov. 2000.
Work in progress.
[3] J. Rosenberg and H. Schulzrinne, "Reliability of provisional
responses in SIP," Internet Draft, Internet Engineering Task Force,
July 2000. Work in progress.
[4] S. Petrack and L. Conroy, "The PINT service protocol:extensions
to SIP and SDP for IP access to telephone call services," Internet
Draft, Internet Engineering Task Force, Feb. 2000. Work in progress.
[5] J. Kuthan and J. Rosenberg, "Firewall control protocol framework
and requirements," Internet Draft, Internet Engineering Task Force,
June 2000. Work in progress.
[6] J. Rosenberg, D. Drew, and H. Schulzrinne, "Getting SIP through
firewalls and NATs," Internet Draft, Internet Engineering Task Force,
Feb. 2000. Work in progress.
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