Internet Communications Using SIP : Delivering VoIP and Multimedia Services With Session Initiation Protocol {2Nd Ed.}

would generate it. The 183 Session Progressis not used. In this scenario,

if the SIP user agent returned an error response (such as 410 The Number is

No Longer in Service), the gateway would be responsible for playing a

suitable announcement for the PSTN caller. In the long term, this will involve

a text-to-speech conversion in which the gateway would speak the error rea-

son phrase. In the short term, the gateway will need to play a prerecorded

announcement, or forward an INVITEto an announcement server, which can

play the announcement.

Many examples of SIP and PSTN interworking are given in the IETF SIP

PSTN Call Flows Best Current Practice (BCP) RFC 3666 [4]. Detailed informa-

tion about mapping between SIP and ISUP can be found in RFC 3398 [5].

SIP Telephony and ISUP Tunneling

SIP internetworking with the PSTN at the signaling level involves a mapping

of message types and parameters from one network to another. For example,

consider the PSTN-to-SIP call in Figure 11.3. The ISUP Initial Address Message

(IAM), or Message 1 in Figure 11.3, is shown in Table 11.1, along with a

description of each field.

Figure 11.2 SIP-to-PSTN call with in-band call progress indicators

SIP User Agent Gateway PSTN Switch Telephone

One way RTP Media One way Speech

1 INVITE

3 100 Trying

Address Complete

Message (ACM)

maps to SIP 183

Session Progress so

that SIP caller hears

ringtone, busytone,

or recorded

announcements.

9 BYE

10 REL

RTP Media Session Two way Speech

2 IAM

4 ACM

5 183 Session Progress

6 ANM

7 200 OK

11 200 OK

No Media Session No Speech Path

12 RLC

8 ACK

190 Chapter 11