SIP servers that normally route all calls are not shown in Figure 11.6 so as to
focus the example on only peer-to-peer third-party call control. In addition,
complete message transaction sequences (such as REFER/200 OK and
INVITE/200/ACK) are grouped and represented by single, two-headed
arrows. This example is based loosely on [13]. The dialer will now use third-
party call control to go through a number of steps to set up a call between the
boss and a customer:- The dialer refers the phone to set up a call to the boss using a REFER
request. The result of the REFERis sent in the NOTIFYresponse. This
occurs after each REFERand enables the dialer to know the exact state
of the call. - The dialer refers the phone to put the boss on hold. This is accom-
plished by sending a REFERto the phone to set the media stream to
sendonlyor inactive. - The dialer refers the phone to place a call to the customer.
- The dialer refers the phone to place the customer on hold.
- The dialer refers the boss’s phone to call the customer. The boss and the
customer now have an RTP voice “call” established. The dialer now
proceeds to get out of the loop. - Dialer refers the phone to send a BYEto the boss.
- Dialer refers the phone to send a BYEto the customer.
- The boss and the customer continue to talk.
Real-life scenarios would look more complex, since we have omitted several
messages in this example for the sake of clarity. There would be extra message
exchanges for presence to display more extensive information about the status
of the phone, and messages with SIP servers to route the calls. In addition, as
discussed in Chapters 9 and 10, the existence of firewalls and NATs will com-
plicate third-party call control.Summary
This chapter has examined SIP telephony, internetworking with the PSTN, and
feature implementations. Even though SIP uses a radically different call model
and structure, basic and enhanced telephony services can easily be imple-
mented in a SIP-enabled network.206 Chapter 11