PDF of this tutorial- Power per channel of the solution expressed in milli-Watts (mW) per channel
- Cost per channel of the solution that includes silicon/hardware, software, and intellectual property licensing costs
- Channel density of the solution expressed in channels per square inch
- System partitioning including packet aggregation and routing
- Software features that define the functionality of the product
- Network management capabilities to address high availability and accountability
Cost, power, and area must be evaluated on a total system basis and must be a function of the features and capabilities supported. Figure 1 shows an example of high density design consisting of the following hardware modules:
- Alarm monitor and control (M&C) module
- Call processing modules
- Public switched telephone network (PSTN) interface modules
- Packet interface modules
- VoP/universal port (UP) modules
- Backplane interface
Figure 1. High-Density VoP Reference Design
The alarm M&C module performs the overall network management for the equipment. This includes configuration on a per-channel basis, status and statistics collection, call record reporting, and alarm processing. The call processing modules perform call establishment and call teardown for the system and performs interworking functions between the PSTN and packet network. Depending on the application and location of the equipment, the following signaling may be performed:
- PSTN telephony signaling
- Signaling system 7 (SS7)
- Integrated services digital network (ISDN)
- TR08
- TR303
- VoP network signaling
- H.323
- Media gateway control protocol (MGCP)
- Megaco
- Session initiation protocol (SIP)
- Asynchronous transfer mode (ATM) broadband local emulation services (BLESs)
Depending on the architecture, call processing may be centralized or distributed with the VoP modules performing lower levels of the signaling protocols.
The PSTN interface modules provide the interface to the PSTN. Traditionally, PSTN interfaces for VoP consisted of T-1 (24 channels) and E-1 (30 channels). High-density VoP systems being designed today typically have multiple DS–3 (28 T-1s or 672 channels) and even multiple OC–3 (2,016 channels) PSTN interfaces as manufacturers offer equipment capable of handling in excess of 100,000 voice channels in a single rack of equipment.
The packet interface modules provide the interface to the packet-switched network. The two most prevalent networks are ATM and Internet protocol (IP). Depending on the application, VoP equipment may be ATM–centric, IP–centric, or support a hybrid of both ATM and IP voice. In many cases, it is important for the equipment to support both voice over ATM (VoATM) and voice over IP (VoP) on a per-call basis to provide interworking between ATM and the IP world. Packet interfaces include OC–n (OC–3, OC–12, etc.) optical interfaces for ATM and packet over SONET (POS) as well as multiple 100 BaseT and Gigabit Ethernet interfaces.
The switch fabric module performs the routing of cells/packets through the system. Line cards fill out the appropriate header information that is used by the switch fabric to direct cells/packets to the appropriate line card/external interface.
As shown in Figure 2, the VoP modules consist of a “farm” of DSPs that perform the actual conversion of the voice streams between the PSTN and packet worlds. In the PSTN–to-packet network direction, the VoP modules receive 64 kbps data streams from the PSTN interface modules and output packets or cells to the packet interface modules. Similarly, in the packet network to PSTN direction, the VoP modules receive packets or cells from the packet interface modules and output 64 kbps streams to the PSTN interface modules. The DSPs are controlled by a “host” processor that is responsible for configuration and software download of the DSPs as well as assisting in call establishment and termination and other network management functions.
In order to concentrate a large number of VoP channels, aggregation logic is required. This logic performs the following functions:
- Aggregates packet streams from multiple DSPs to the backplane/packet network interface
- Routes incoming packets from the backplane/packet network interface to the appropriate DSP
- Provides a standard interface to the backplane/packet network interface
- Filters network management and call setup/teardown information to a host processor
There are many different backplane interfaces that are used in systems such as these. Most typical are PCI and cell bus variants as well as POS. Time division multiplexing (TDM) samples from the PSTN can be relayed over an H.110 TDM bus or the PCM samples can be encapsulated in ATM cells to be sent over the same cell bus that is used for packet traffic.
Figure 2. VoP High Density Module
Figure 3. VoP Aggregation Logic
Software is a critical ingredient of high-quality VoP systems. There are many features that must be implemented for carrier-class systems. The most important software features include the following:
- Echo cancellation
- Voice compression
- Packet play-out software
- Tone processing
- Fax and modem support
- Packetization
- Signaling support
- Network management
Echo Cancellation
One of the keys to high-quality VoP is having a hardened line echo canceller that can properly cancel echo, which is present even in a conventional POTS network. In this type of network, echo is acceptable because delay is less than 50 milliseconds, and the echo is masked by the normal side tone that every telephone generates. Echo becomes a problem in packet networks because the delay is almost always greater than 50 milliseconds, thus requiring echo-cancellation techniques as part of the VoP solution. The ITU defines echo cancellation performance requirements. The original standard for echo cancellers was ITU Recommendation G.165; a more stringent set of requirements is provided in ITU Recommendation G.168. These standards provide a series of objective performance tests but do not describe how to implement an echo canceller, nor do they address the subjective performance of an echo canceller.
A good echo canceller must have the following attributes:
- It removes echo well. This includes removing echo at the start of a call as well as preventing any form of echo during a call.
- It handles double talk (both sides talk simultaneously) well. This includes not clipping the voice at the beginning or end of a double-talk voice spurt.
- It handles background noise well. This includes handling high background noise and variable background noise.
- It exceeds G.165/G.168 and provides support for future (more stringent) ITU EC standards such as G.168-2002.
- It is field proven. Note: Compliance with G.165/G.168 alone is no guarantee that the EC will work properly in real-life situations.
- It provides fast convergence time, low residual echo (depth of convergence), reliable detection of double-talk without divergence or clipping, and handles background noise and narrowband signals well.
- It supports up to 128 millisecond tail (often specified for carrier-class solutions) including support for multiple reflections over the entire 128 msec tail.
- It is capable of dynamically tracking echo path changes and is required for conferencing, call transfers, permanent off-hook connections, and to support redundancy.
- It behaves properly in the presence of 4-wire connection and low hybrid attenuation.
- It has built-in configurability and instrumentation.
Voice Compression
G.711 PCM (64 Kbps) is being most widely deployed for very high density VoP solutions. This is enabled by having “bandwidth to burn” in the infrastructure and the fact that PCM requires less processing power per channel than for low bit-rate (LBR) compressed voice results in higher densities at a lower cost from an equipment perspective. LBR compressed voice is required for broadband access equipment where bandwidth to the infrastructure is constrained, such as the following:
- Derived voiceover–cable modem and fixed wireless applications where there is a shared medium from the subscriber device to the head-end
- Derived voice channels over DSL applications where the subscriber line is a long distance away from the CO
- Mobile-to-mobile cellular calls where voice is already compressed, and it is undesirable to perform transcoding (degrades voice quality and adds delay)
- Emerging Internet voice portal market
- Internet audio content is stored and/or transmitted as compressed data compatible with PC programs
- Wideband voice compression where new algorithms fulfill the promise of packet voice quality better than the PSTN
The following are important considerations for voice compression:
- Passes all ITU test vectors
- Configurable packetization for maximum flexibility
- Proprietary VAD required for voice CODECs that do not support an integral VAD
- Adaptive comfort noise generation (CNG) in conjunction with VAD
- No degradation when all channels are active
Packet Play-Out
Packet play-out addresses the effects of network impairments on the voice. Impairments include lost packets, and delay of packets, including variable delay that distorts the timing sequence of the original voice. Therefore, it is essential for high-quality VoP to have packet play-out algorithms. These algorithms should do the following:
- Compensate for packet loss, delay, and jitter
- Be adaptive for lowest delay
- Reside in DSP for system scalability
- Be highly configurable and provide comprehensive network management statistics
Tone Processing
Tone processing is essential for call setup and termination as well as handling in call user functions such as accessing voice mail, making credit card calls, etc. The following are key elements for high density systems:
- Reliable tone detection (no false detects, no failure to detect)
- Early detection to minimize delay and to prevent in-band tone leakage that can lead to false tone generation at the remote end
- Different detection requirements based on network application and system architecture: dial digits, fax detection, modem detection, and call progress tones
- Support for bidirectional tone detection and generation in cases where the customer premises equipment (CPE) does not perform these functions
Fax and Modem Support
One of the key features of a VoP System for carrier-class applications is to fully emulate the PSTN, e.g., Class-5 switch replacement. Therefore, it is essential that these systems handle fax and modem in addition to voice. Fax and modem can be handled by a technique known as “PCM up-speed” where the VoP system, upon detection of fax or modem, forwards all data for a given channel as a 64 kbps transparent PCM stream between the two end points. While this works well for networks with no packet loss or excessive delay/jitter, networks (particularly VoP networks) experience occasional packet loss (less than 1 percent) that can cause the modems to retrain or even call failure. In cases where PCM up-speed is inadequate, the system must support fax and modem relay.
Fax relay provides reliable real-time fax service between two analog fax machines over a packet network. The VoP equipment at both ends of the packet network spoofs the analog fax machines such that they operate as if directly connected over a PSTN connection. The VoP equipment performing fax relay functions must handle the effects of network delay, jitter (variable delay), and lost packets while preventing the fax machines from timing out. Standards protocols such as T.38 and AAL–2 exist for interoperability between equipment vendors. Proprietary techniques are often used to improve the interoperability between different fax machines that are subjected to long delay and other packet-network effects. Fax relay consists of the following functions:
- Fax modem pumps: V.17, V.29, V.27ter, V.21
- Fax relay protocols: T.38 (TCP/IP), AAL–2 (ATM)
- Fax machine spoofing protocols: proprietary
Currently, there are no interoperability standards for modem relay, but standards are being proposed. Modem relay consists of the following functions:
- Modem pumps: e.g., V.90, V.34, etc.
- Modem relay protocol: negotiation, flow control, error control
Packetization
The following packet encapsulation is performed in the DSPs to facilitate scalability and flexibility:
- VoP (RTP/RTCP)
- AAL–2
- AAL–1xN (for videoconferencing streams)
These should be supported on a per-channel basis to support hybrid ATM/IP networking equipment. Another important feature is something known as network channel switching. This is the ability to route from packet network to packet network. The routing can include the ability to transcode the voice payload and/or change encapsulation format, as in the following examples:
- VoP (RTP) <–> AAL–2
- G.726 <–> G.729AB
Signaling
Signaling support is an essential element of the DSP software. Features include the following:
- Full tone detection and generation capabilities: DTMF, MF R1/R2, SS7 COT, call progress tones, bidirectional tone processing
- Channel associated signaling (CAS) support: CAS bit processing
- Common channel signaling (CCS) support: HDLC, MTP1 (SS7)
- CAS and CCS support in DSP to off-load the host for higher scalability
- Play-out of service announcements (TDM or packet network direction)
Network Management
Fundamental to any communications system is the ability to discover, isolate, and remedy problems as quickly as possible to minimize or eliminate the degree to which users are impacted.
- Configuration on per-channel basis including set-able country-code specific information
- Per-channel statistics and status reporting
- Per-channel real-time trace and diagnostics capabilities
- Bellcore test line support for diagnostics
- Redundancy support
Figure 4. High-Density VoP Software Architecture
Figure 4 shows the architecture for high density VoP software. The software should be designed to minimize delay and maximize scalability. This includes the following:
- Efficient and adaptive algorithms
- Header encapsulation in VoP device, (RTP, AAL–1, AAL–2) on a per-channel basis
- Low-latency implementation
- Features to off-load “host” processor to drive overall channel density
