VOIP design for added value
01 January 2007
The age of VoIP (Voice-over-Internet-Protocol) is here, bringing together telephony and data communications to provide packetised voice and fax data streamed over low-cost Internet links.
VoIP technology has the potential to revolutionise the way people communicate, whether they’re at home or at work, video-enabled or just plain audio-minded.
The VoIP challenge to the embeddedsystem designer is to choose a processing solution that is cost-effective, easy to deploy, and scaleable in performance across market spaces. A ‘sweet-spot’ embedded-solution approach is to design with a platform that can implement a low-channel-count basic VoIP solution, yet retain capacity for valueadded capabilities and services, like video, music, imaging, and system control.
There are various features that VoIP brings, for instance incoming phone calls on the PSTN (public switched telephone network) can be automatically rerouted to a user’s VoIP phone, as long as it is connected to a network node. This arrangement has clear advantages over a global-enabled mobile phone, since there are no roaming charges involved. From the VoIP standpoint, the end user’s location is irrelevant, it is simply seen as just another networkconnection point. This is especially useful where WLANs (wireless local area networks) are available. IEEE-Standard- 802.11-enabled VoIP handsets allow conversations at worldwide Wi-Fi hotspots without the need to worry about mismatched communications infrastructure and transmission standards.
VoIP extends to other forms of data-based communication. Once data is digitised and packetised, the nature of the content does not much matter. The VoIP infrastructure facilitates a new set of networked real-time applications, such as video conferencing, multicasting and electronic whiteboards.
A closer look
Figure 1 shows key components of a VoIP system, the signalling process, the encoder/decoder, the transport mechanism, and the switching gateway. The signalling process involves creating, maintaining, and terminating connections between nodes.
In order to reduce network bandwidth requirements, audio and video are encoded before transmission and decoded during reception. The compressed packets move through the network governed by one or more transport protocols. A switching gateway ensures that the packet set is interoperable at the destination with another IP-based system or a PSTN system. At its final destination, the packet set is decoded and converted back to an audio/video signal, at which point it is played through the receiver’s speakers and/or display unit.
The Open Systems Interconnection sevenlayer model (see figure 2) specifies a framework for networking. If there are two parties to a communication session, data generated by each starts at the top, undergoing any required configuration and processing through the layers, and is delivered to the physical layer for transmission across the medium. At the destination, processing occurs in the reverse direction, until the packets are finally reassembled and the data is provided to the second user.
Protocol sessions
The first requirement in a VoIP system is a session-control protocol to establish presence and locate users, as well as to set up, modify, and terminate sessions. There are two protocols in wide use today. Historically, the first of these protocols was H.323, but SIP (Session Initiation Protocol) is becoming the standard.
The signalling protocols (H.323 or SIP) are responsible for configuring multimedia sessions across a network. Once the connection is set up, media flows between network nodes are established by using one or more data-transport protocols, such as UDP (User Datagram Protocol) or TCP (Transmission Control Protocol).
Sending media data directly over a transport protocol is not very efficient for real-time communication. Because of this, a media transport layer is usually responsible for handling this data in an efficient manner. RTP provides delivery services for real-time packetised audio and video data. It is the standard way to transport real-time data over IP networks. The protocol resides on top of UDP to minimise packet header overhead, but at a cost. There is no guarantee of reliability or packet ordering. At the top of the VoIP stack are protocols to handle the actual media being transported. There are potentially quite a few audio and video codecs that can feed into the media transport layer.
Blackfin in VoIP
Traditional VoIP embedded solutions have used two processor cores to provide VoIP functionality. However, Blackfin processors provide a convergent solution in a unified core architecture that allows voice and video signal processing concurrent with RISC MCU processing to handle network- and userinterface demands. This ability to offer full VoIP functionality on a single convergent processor provides for a unified software development environment, faster system debugging and deployment, and lower cost.
The ADSP-BF537 Blackfin processor family provides the degree of integration and performance, with low power consumption, necessary for VoIP deployment. It features multiple integrated serial ports (for glueless connection to audio analogue-to-digital (A/D) and digital-to-analogue (D/A) converters), an external memory controller, a parallel peripheral interface (PPI) for LCD or video encoder/decoder connectivity, and a 10/100BaseT Ethernet MAC.
A complete communication channel uses less than 75MIPS of the processing bandwidth. With ADSP-BF537 performance at up to 600MHz, there is plenty of available processor ‘horsepower’ to spread across a VoIP product portfolio, as features such as multimedia compression or decompression become necessary. In contrast, competing dedicated VoIP choices are typically performance-limited and offer little or no ability to add features or differentiation.
For VoIP applications, designs based on devices like Blackfin can be used to target high-quality, low-channel-count VoIP solutions, with processing headroom to accommodate added features such as music, video, image transport, and overall system control.
VoIP offerings
There are a number of open source solutions and reference designs available for VoIP. For example, there is the open-source software based on ìClinux, the embedded version of the popular GNU/Linux OS. One such General Public License (GPL-licensed) IP-phone package, called Linphone has been ported to ìClinux for Blackfin processors, allowing the reference design to communicate with any SIP-compatible endpoint. In a public network with the proper SIP servers and gateway infrastructure, this system can even be used to connect to phones on a PSTN node. For voice encoding and decoding, the current implementation supports: G.711 (A-law and m-law), GSM (Global System for Mobile Communications), and the Speex audio compression format.
The Blackfin Linphone reference design uses Linux TCP/IP networking stack, which includes necessary transport and control protocols, such as TCP and UDP. The G.711 and GSM codec implementations comprise a graphical user interface (GUI) for desktop PCs and a simple command-line application for non-graphical embedded systems.
There is also an implementation of an RTP stack, the oRTP, developed for Linphone and released under the LGPL licence. A threadsafe implementation of the SIP is the oSIP, which is used with Speex, the open-source reference of the eponymous codec. Optimisations to the fixed-point Speex implementation have been contributed back to the mainline code branch.
Working models
The Fusion Voice Gateway (see figure 2) is a Voice Gateway/Terminal Adapter reference design from Unicoi Systems. With router functionality and full-featured SIP telephony running on a single-core Blackfin processor, the design allows for quick time-to-market for terminal adapters. The reference design features robust functionality, including G.168 echo cancellation and multiple G.7xx voice codecs. It also includes full-featured telephony and router functionality by combining an Internet router, a four-port Ethernet switch and VoIP gateway functionality.
The Blackfin BRAVO VoIP and Videophone reference designs enable OEMs to build feature-rich, high-performance, low-cost VoIP desktop phones, videophones, and telephone adapters. The designs include the complete suite of software for VoIP applications, controlled by a set of application programme interfaces (APIs) for customisation and control of core system functions.
The BRAVO Broadband Audio/Video Communications reference design provides up to 30frame/sec of common intermediate format (CIF) color video, including support for ITU-standard H.263 and H.264 video codecs, picture-in-picture, high-resolution graphics with overlay, alpha and chroma keying, and antiflicker filtering.
Co-authored by DAVID KATZ, TOMASZ LUKASIAK, RICK GENTILE and WAYNE MEYER, Analog Devices.
Contact Details and Archive...
Related Articles...
Most Viewed Articles...