Third party call control (3pcc) презентация

Security
Instruction : Prof. Schulzrinne

a call in which communication is actually between other parties.
use the mechanisms specified within the Session Initiation Protocol (SIP).
Examples of use:
Operator services
conferencing

wishes to create a Session between two other user agents.
issue either its own or other’s party “forged” From address.
does not carry the media stream sent over the call channels.
often called back-2-back user agent

session is created by a third party, referred to as the controller.
benefits and drawbacks to each of these flows

no SDP

INVITE offer1

200 OK answer1

ACK

ACK answer1

RTP Media

Timeline

manipulation of the SDP by the controller
works for any media types supported by both endpoints.
Cons:
timeout problem

Media

4

ACK

8

200 OK sdp2

9

ACK

bh= Black hole
connection address=0.0.0.0

Flow II-Alternative flow with bh sdp

are immediately ACKed.
no timeout and message inefficiency problems of flow 1.
Cons:
the controller need know the media types to be used for the call
bh sdp sent and bh sdp response
Infinite loop of re-INVITEs

offer2

ACK answer2

INVITE offer2’

Timeline

RTP Media

3

ACK answer1(bh)

7

200 answer2’

9

ACK

controller need not guess the media that will be used by the participants.
Cons:
Controller need to perform SDP manipulations.
Reorder and trim SDP X is done by controller, so that the media lines match up with those in some other SDP, Y.
The controller need to detect and terminate the call with different codecs and media line.
the flow is far more complicated than Flow I

SDP

200 OK offer2

ACK answer2

INVITE offer2’

Timeline

RTP Media

3

ACK

7

200 answer2’

9

ACK

Flow IV - a variation on Flow III

the media manipulations by controller is more simply.
Cons:
user A will be alerted without any media having been established yet.
If there is no media in common,
user annoyance and possibly resulting in billing events.

B is actually an automata that will answer the call immediately.
media servers, conferencing servers, and messaging servers
Flow II
SHOULD NOT be used

it provides no additional benefits over Flow IV.
Flow IV
For calls to unknown entities, or to entities known to represent people, it is RECOMMENDED

B would fail。
User B is busy
no media in common
the request time out
Solution:
the controller send a BYE to A.
Reason header--carries the status code from the error response.

no SDP

180

INVITE offer2

3

ACK

7

491

8

ACK

180: Ringing
491: Request pending

Timeline

create a new BYE to finish the session
Forward the re-INVITE receive from one participants

OK

6

INV no SDP

5

ACK

8

INV offer3’

11

ACK answer3

established.
Before the call is setup, the msg(tones or anouncements) that about this call

is needed by the controller
Controller can cut the session if it is necessary(no overlapping codec or media)
There may be another possible the precondition is needed by the participant

no SDP

183 offer2
Des=sendrecv
Conf=recv
Cur=none

Timeline

UPDATE offer2’
Des=sendrecv
Conf=recv
Cur=none

8

200 UPDATE answer2’
Des=sendrecv
Conf=recv
Cur=none

9

PRACK answer2
Des=sendrecv
Conf=recv
Cur=none

10

200 PRACK

11

reservation

183 Session in Progress

offer4’
Des=sendrecv
Conf=recv
Cur=sendrecv

19

200 UPDATE answer4’
Des=sendrecv
Conf=recv
Cur=sendrecv

Timeline

OK

INV offer2’

ACK’

6

Timeline

3

INV offer1 no media

4

INV answer1 no media

9

200 answer2

10

ACK answer2

Click-to-dial

with an address of 0.0.0.0.
Re-INVITE requests that change the port to which media should be sent o Re-INVITEs that change the connection address
Re-INVITEs that add a media stream
Re-INVITEs that remove a media stream (setting its port to zero)
Re-INVITEs that add a codec amongst the set in a media stream
SDP Connection address of zero o Initial INVITE requests with a connection address of zero
Initial INVITE requests with no SDP
Initial INVITE requests with SDP but no media lines
Re-INVITEs with no SDP

Best Current Practices for Third Party Call Control (3pcc) in the Session Initiation Protocol (SIP)”, RFC3725, Apr, 2004
T. Chiang, V. Gurbani, J.Reid,”The Need for Third-Party Call Control”,Lucent Tech., 2002