Detailed Call Walk-Through

Earlier in this chapter, we presented an ISUP message flow in order to illustrate the exchange of messages to establish and release an ISUP call. Now that we have discussed more of the ISUP details, we will build on that illustration. This section provides more detail about the call processing that was driven by the ISUP message events used in the earlier example. Although this chapter's primary focus is the ISUP protocol, it is important to understand how ISUP is applied in its normal domain of trunk call processing.

Call Setup

Refer back to Figure 8-3, where a call originates from a line at SSP A and terminates to a line at SSP B over an interexchange ISUP trunk. When call processing has completed translations of the called number at SSP A, the translations' results indicates that the call requires routing to an interexchange trunk group. The provisioned signaling type for the selected trunk group determines whether ISUP signaling or some other signaling, such as Multifrequency (MF), is used. When the signaling type is determined to be ISUP, the trunk circuit to be used for the outgoing call is reserved for use.

The SSP populates the IAM with information about the call setup, such as the CIC, CdPN, Call Type, CgPN, and PCM Encoding scheme. The IAM information is placed in the User Data field of the MTP3 SIF. The MTP3 information is populated based on the SS7 network information that is associated with the selected trunk group. As previously noted, each switching exchange contains a provisioned association (usually static) between routesets and trunkgroups. The IAM is then transmitted onto a signaling link toward the destination identified in the message by the DPC. If quasi-associated signaling is used, the message's next-hop node is an STP that will route the message to the intended SSP. If associated signaling is used, the IAM is transmitted directly to the SSP that is associated with the trunk being set up. SSP A starts timer T7, which is known as the network protection timer, or the awaiting ACM timer, to ensure that an ACM is received in response to the IAM.

When SSP B receives the MTP3 message, it recognizes it as an ISUP message by the SIO's Service Indicator bit. Then the message is passed to ISUP for processing, during which it extracts the message information. An IAM indicates a request to set up a call so SSP B enters the call processing phase for a trunk origination. The CdPN and Calling Party Category fields provide key pieces of information from the IAM for SSP B to complete number translations for this simple call.

In this example, the number translates to a subtending line of SSP B, which checks the line to determine whether it is available. An ACM is built and sent to SSP A, notifying that the call can be completed and is proceeding. At this point, the speech path in the backward direction (from SSP B to SSP A) should be cut through to allow the ring-back tone to be sent over the bearer channel from the terminating exchange to the originating exchange. This indicates that the terminator is being alerted.

Ringing is now applied to the terminating set, while ring back occurs at the originating set. Answer timing is usually applied at the originating switch to limit the amount of time an originator waits for answer.

When the terminating subscriber goes off-hook, an ANM is sent back to the originator to indicate that an answer has occurred. By this point, the voice path should be cut through in the forward direction to allow the conversation to take place. Note that the voice path can be cut through before receiving the ANM, but it must be cut through no later than the ANM. The call is now in the active, or talking, state. This is often a point of interest for billing procedures that require capturing the time at which a call conversation begins. For an ordinary call, no further signaling messages are exchanged for the duration of the conversation. When either of the parties goes on-hook, it initiates signaling for the release of the call. The following section discusses Call release.

Call Release

When either the originating or terminating subscriber goes on-hook, it signals an attempt to disconnect the call. In Figure 8-3, the originator at SSP A goes on-hook. SSP A recognizes the signal to disconnect the call and sends a Release message (REL) to SSP B. SSP B responds by sending a Release Complete message (RLC) as an acknowledgement. The trunk member is freed and placed back into its idle queue to be used for another call.

Terminal Portability

The ITU defines terminal portability in Q.733.4 for allowing the called or calling party to hang up a phone and resume a conversation at another phone that is connected to the same line. When the two parties are connected over an inter-exchange ISUP trunk, suspend and resume messages are used to maintain the trunk connection until the on-hook party has gone off-hook. Terminal portability requirements for the called party exist in many countries; however, terminal portability for the calling party is not supported as often. ANSI networks do not support terminal portability for the calling party.