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Chapter 2

Signaling and Call Progress

T1 Robbed bit

The analog world used the reversed battery signaling and the E & M (4 wire system for PBX extensions) signaling to indicate seizures, acknowledgements and even to do pulse dialing. The E & M scheme uses two extra leads for the signaling information the ‘E’ lead, or the Ear lead, the receiving lead and the ‘M’ lead or Mouth lead, the transmitting lead. When an extension goes off hook, the ‘M’ lead’s potential goes to Power Supply and the ‘E’ lead to ground. When back to On Hook ‘M’ goes to ground and ‘E’ goes open. To emulate the E & M leads, the digital approach had to provide two bits, they were called the ‘A’ and ‘B’ bits. Since no more bits could be added to the stream, the designers decided to “steal” a couple of bits from the data in a super frame:

the 8^th bit of every time slot in the 6^th frame of every super frame will be regarded as the signaling bit ‘A’ of that time slot; 

the 8^th bit of every time slot in the 12^th frame of every super frame will be regarded as the signaling bit ‘B’ of that time slot. 

This would hardly have any effect on the sound and would provide for the necessary signaling.

Digital PBX’s use signaling protocols involving both bits but most of the inter-switch communications use only the ‘A’ bit or ‘A’ and ‘B’ in a mirror fashion.

Handshakes for Call Setup

The idle state value for the signaling bits is something that is part of the handshake protocol and can be set in any of the four different possibilities but the general usage is that idle => A = B = 0.

The simplest handshake is the one called “Immediate Start”. Say Switch 1 is setting up a call on Switch 2. Switch 1 sets it Bit ‘A’ to 1, this meaning seizure for Switch 2, which gets ready to receive the dialing digits. After a guard time, Switch 1 sends the digits. Switch 2 attempts connection, if successful it raises its ‘A’ bit meaning connection acknowledge or Answer Supervision. If called party hangs up, Switch 2 loses connection and lowers it ‘A’ bit. When caller hangs up, Switch 1 set its ‘A’ bit to zero, call is over and Switch 2 lowers it ‘A’ bit too.

The Wink or Flash is a short signal (100 to 250 ms) used to trigger some functions in the PBX’s and switches. Digital telephony also inherited the Wink concept. The Wink allows for a faster handshake known as “Wink Start”. The Wink is used to acknowledge the seizure, it sends this message “I’m ready for the dialing digits, don’t wait the guard time”.

Dialing

In the T1 robbed bit world dialing may be performed in three ways:

·         Pulse Dialing (pulsing the signaling bits in the same way the loop current was pulsed)

·         DTMF (Dual Tone Multi Frequency)

·         MF (Multi-Frequency)

DTMF digit are characterized by a double tone pulse of 50 to 100 ms and their frequencies go according to the following table:

Low T    

MF is mostly used for inter-switch communication in the US.

 When setting up an inter-switch link, the dialing mode must match and this is a frequent source of problems during setup.

Call Progress Tones

The T1 carrier system inherited the same call progress indicator used in the analog world, the Call Progress Tones.

 To this we must add the SIT tones (Special Information Tones) that precede a recorded message from the CO (Central Office).  The SIT’s are three precisely pitched and timed tones that machines can detect as an Operator Intercept and not as a human answer to a call. When there are no SIT tones, case that is quite common in international calls, wrong “answer supervision” may be issued by the receiving switch and a caller may be unfairly charged for an uncompleted call. 

 The E1 Carrier System

An E-1 digital trunk operates at 2.048 Mbps divided into 32 time slots with each time slot operating at 64 Kbps. These 32 time slots include:

· 30 time slots available for up to 30 voice calls

· one time slot dedicated to carrying frame synchronization information (timeslot 0)

·  one time slot dedicated to carrying signaling information (time slot 16)

·     the international bit occupies the most significant bit (bit position 7) in time slot 0 of each frame

·      the lower 6 bits of every even frame carry the framing pattern 0011011

·       the national bits occupy bit positions 0 through 4 of time slot 0 of every odd frame

 Time slot 16 in frame 0, carries:

MSB                                                                                            LSB

X - Extra Bits, Used for Multi-frame Synchronization

Y - Distant Multi-frame Alarm Bit

 The rest of the 15 frames in the super frame carry the signaling information for the time slots, two per frame as follows:

 Time Slot 16

R2 MF

The method sending signaling information within the band where voice is being carried is called Channel Associated Signaling (CAS) an example of which is the R2 MF protocol. CAS service is available in Europe and in parts of Asia and South America. The Conference of European Postal and Telecommunications administration (CEPT) defines how a PCM carrier system in E-1 areas will be used.. One wonders why having a full 64Kbit in channel 16 just for signaling someone needs to send in-band information, well... probably there are historic reasons for this, ISDN and SS7 protocols are newer. E-1 CAS service may carry national and international signaling bits set in time slot 0, at least its using the other channel.:

For each E-1 CAS call, signaling information is sent to the local CO and then to each successive CO until the destination CO is reached. The destination CO attempts to connect to the called party. Concurrently, the destination CO sends back signaling information representing the condition or status of the called party’s line. This signaling information passes through the network as audio tones. R2 MF signaling is the international standard for conveying call status using these audio tones. However, the number of tones used, the frequency combinations used, and the adherence to the R2 standard can vary from country to country.

R2 MF inter-register signaling uses forward and backward compelled signaling. Simply put, with compelled signaling each signal is sent until it is responded to by a return signal, which in turn is sent until responded to by the other party. Each signal stays on until the other party responds, thus compelling a response from the other party.

Reliability and speed requirements for signaling systems are often in conflict, the faster the signaling, the more unreliable it is likely to be. Compelled signaling provides a balance between speed and reliability because it adapts its signaling speed to the working conditions with a minimum loss of reliability.

 The R2 MF signal is composed of two significant events, tone-on and tone-off. Each tone event requires a response from the other party. Each response becomes an acknowledgement of the event and an event for the other party to respond to.

Compelled signaling must always begin with a Group I forward signal.

 ·         The CO starts to send the first forward signal.

·         As soon as the CPE recognizes the signal, it starts to send a backward signal that serves as an acknowledgement and at the same time has its own meaning.

·         As soon as the CO recognizes the CPE acknowledging signal, it stops sending the forward signal.

·         As soon as the CPE recognizes the end of the forward signal, it stops sending the backward signal.

·         As soon as the CO recognizes the CPE end of the backward signal, it may start to send the next forward signal.

 The CPE responds to a tone-on with a tone-on and to a tone-off with a tone-off. The CO responds to a tone-on with a tone-off and to a tone-off with a tone-on.

Refer to the following figures for more information:

There are many details about this protocol that we won’t go into but it is certainly faster than  DTMF or MF T1 robbed bit and a lot more information about the call, the called party and the caller can go back and forth. These advantages come with quite a big implementation price tag difference though, Since seizures and signaling, all come as compelled tone interchanges, you need a voice resources attached to every channel. The hardware for a 4 E1 wink start OmniBox system could go for say $11000 while an R2 MF will require more than a $32000 investment. 

 ISDN PRI

The Integrated Services Digital Network (ISDN) is a digital communications network capable of carrying all forms (voice, computer, and facsimile) of digitized data between switched endpoints. ISDN may use D4 SF, ESF and E1 framings but what distinguishes ISDN from earlier protocols is signaling.

 Not only that signaling is done in an out-of-band fashion (E1 accomplished that too) making all channels potentially “Clear”, it is that the quality of signaling is changed to one that can convey a lot more information at a speed far beyond that of the fastest R2 MF.

ISDN was implemented for the T1 carrier system by using the 24^th timeslot, the first 23 are called “B channels” (Bearer channels) while the 24^th is the “D channel” (Data channel). The implementation allows the use of one timeslot for up to 8 T1, this is called NFAS (Non Facility Associated Signal). With the E1, no voice channel needs to be sacrificed for signaling, since it was already out of band in the 16^th timeslot. In both cases there is a 64Kbts channel for data information.

 With ISDN, a call can be setup with information about the caller (ANI) and the called number (DNIS) in the millisecond range. A single DTMF digit, in its shortest version, is 50 ms plus a 50 ms space, a ten digit number requires a whole second. Is not just faster, is 1000 times faster, such a huge difference brings up a totally different ball game. It takes only a few milliseconds for a switch to get the dialed digits, analyze them and, if there are no available channels to the requested destination, it may choose not to “accept” the call. I can even tell the reason for not accepting it. Now the seizing switch still has plenty of time to reroute the call to alternative routing options. This opens the concept of a channel state beyond the narrow Off hook/ On Hook scheme, there is a whole set of possible states to a call now.

 ISDN Call Control States

Each ISDN call that is received or generated by an application is processed through a series of call control states. Each state represents the completion of certain tasks and/or the current status of the call. The following table describes the ISDN call control states, based on standard Q.931 (Layer 3).

 Call Control States

 Here is a sequence for an inbound call:

Event:     Seizure   Accept   Connect                Caller hangs-up    Call is dropped     Call is released

State:      Offered   Accepted Connected          Disconnected       IDLE                       NULL

 Now an Outbound

Event:     Seizure   Ringing  Answer                 Remote hangs-up Call is dropped     Call is released

State:      Dialing   Alerting  Connected            Disconnected       IDLE                       NULL

 When a call is dropped a code for the cause of dropping the call can be sent back to the other end, below a table with the coded causes.

 This is shown only to illustrate the superiority of ISDN technology.

 SS7 Protocol

SS7 has very much the same capabilities as the ones described above but it may use a totally separate line for data information. That data line may be a full T1 a V.35 or any other fast serial link. This link may convey signaling data for many T1’s, and it is today the carrier’s favorite protocol. SS7 is not just a link but a whole network concept.

Signaling Points

Each signaling point in the SS7 network is uniquely identified by a numeric point code. Point codes are carried in signaling messages exchanged between signaling points to identify the source and destination of each message. Each signaling point uses a routing table to select the appropriate signaling path for each message.

There are three kinds of signaling points in the SS7 network:

·        SSP (Service Switching Point)

·        STP (Signal Transfer Point)

·       
SCP (Service Control Point)

SS7 Signaling Points

SSPs are switches that originate, terminate, or tandem calls. An SSP sends signaling messages to other SSPs to setup, manage, and release voice circuits required to complete a call. An SSP may also send a query message to a centralized database (an SCP) to determine how to route a call (e.g., a toll-free 1-800/888 call in North America). An SCP sends a response to the originating SSP containing the routing number(s) associated with the dialed number. An alternate routing number may be used by the SSP if the primary number is busy or the call is unanswered within a specified time. Actual call features vary from network to network and from service to service.

 Network traffic between signaling points may be routed via a packet switch called an STP. An STP routes each incoming message to an outgoing signaling link based on routing information contained in the SS7 message. Because it acts as a network hub, an STP provides improved utilization of the SS7 network by eliminating the need for direct links between signaling points. An STP may perform global title translation, a procedure by which the destination signaling point is determined from digits present in the signaling message (e.g., the dialed 800 number, calling card number, or mobile subscriber identification number). An STP can also act as a "firewall" to screen SS7 messages exchanged with other networks.

 Because the SS7 network is critical to call processing, SCPs and STPs are usually deployed in mated pair configurations in separate physical locations to ensure network-wide service in the event of an isolated failure. Links between signaling points are also provisioned in pairs. Traffic is shared across all links in the linkset. If one of the links fails, the signaling traffic is rerouted over another link in the linkset. The SS7 protocol provides both error correction and retransmission capabilities to allow continued service in the event of signaling point or link failures.

There are recent solutions that convert SS7 data stream to that ISDN PRI which is supported by OmniBox systems. The Technology is called DSC for Digital Signaling Converter from Data Kinetics.

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