Survey on Streaming

Cisco's unified communications offerings are of the most interest to respondents of SearchVoIP.com's recent VoIP and unified communications survey, completed earlier this month.

Almost 82% of survey respondents said Cisco has the strongest and most interesting story and product offerings when it comes to unified communications (UC). Cisco was followed by Avaya, 48%; Microsoft, 44%; and Nortel, 38%. Rounding out the top ten were Alcatel-Lucent, IMB/Lotus, Siemens, 3Com, Polycom and ShoreTel.

The survey queried nearly 500 respondents in various roles within their organizations -- including IT/network/systems operations staff, mid-level IT managers, network engineers, consultants, telecommunications managers and other IT staff members -- about their plans for UC and VoIP.

Despite Cisco's claiming the top spot among respondents, many indicated that they are confused about Cisco's and Microsoft's positions in the UC market. The largest group, 30%, said they don't know where each of the major UC vendors fits. Just over 27% noted that Cisco and Microsoft offer different approaches and compete with each other, while 21% said the two vendors offer different approaches but maintain a partnership. In addition, 16% said Cisco and Microsoft offer similar approaches but compete with each other, while 6% said they believe the pair offer similar approaches and are partners.

When asked where they procure UC solutions, 29% of respondents said they go directly to a single manufacturer or vendor. A similar number, 28%, buy one solution through VARs, resellers or integrators. And 19% buy directly from multiple manufacturers or vendors.

Unicast

Unicast is communication between a single sender and a single receiver over a network. The term exists in contradistinction to multicast, communication between a single sender and multiple receivers, and anycast, communication between any sender and the nearest of a group of receivers in a network. An earlier term, point-to-point communication, is similar in meaning to unicast. The new Internet Protocol version 6 (IPv6) supports unicast as well as anycast and multicast.

Multicast

Multicast is communication between a single sender and multiple receivers on a network. Typical uses include the updating of mobile personnel from a home office and the periodic issuance of online newsletters. Together with anycast and unicast, multicast is one of the packet types in the Internet Protocol Version 6 (IPv6).
Multicast is supported through wireless data networks as part of the Cellular Digital Packet Data (CDPD) technology.

Multicast is also used for programming on the MBone, a system that allows users at high-bandwidth points on the Internet to receive live video and sound programming. In addition to using a specific high-bandwidth subset of the Internet, Mbone multicast also uses a protocol that allows signals to be encapsulated as TCP/IP packet when passing through parts of the Internet that can not handle the multicast protocol directly

Streaming video

Streaming video is a sequence of "moving images" that are sent in compressed form over the Internet and displayed by the viewer as they arrive. Streaming media is streaming video with sound. With streaming video or streaming media, a Web user does not have to wait to download a large file before seeing the video or hearing the sound. Instead, the media is sent in a continuous stream and is played as it arrives. The user needs a player, which is a special program that uncompresses and sends video data to the display and audio data to speakers. A player can be either an integral part of a browser or downloaded from the software maker's Web site.
Major streaming video and streaming media technologies include RealSystem G2 from RealNetwork, Microsoft Windows Media Technologies (including its NetShow Services and Theater Server), and VDO. Microsoft's approach uses the standard MPEG compression algorithm for video. The other approaches use proprietary algorithms. (The program that does the compression and decompression is sometimes called the codec.) Microsoft's technology offers streaming audio at up to 96 Kbps and streaming video at up to 8 Mbps (for the NetShow Theater Server). However, for most Web users, the streaming video will be limited to the data rates of the connection (for example, up to 128 Kbps with an ISDN connection). Microsoft's streaming media files are in its Advanced Streaming Format (ASF).

Streaming video is usually sent from prerecorded video files, but can be distributed as part of a live broadcast "feed." In a live broadcast, the video signal is converted into a compressed digital signal and transmitted from a special Web server that is able to do multicast, sending the same file to multiple users at the same time.

Streaming Conventions and Architectures

The ITS Streaming Video Service supports MPEG, MP3, QuickTime, Real, and Windows Media for video-on-demand. MPEG-2 is only available for on-campus viewers (because of bandwidth requirements). Webcasting is currently being offered in Real and QuickTime.

Which convention to select depends on several factors, including audience location, what streaming players your audience is most comfortable using, and what tools you are most comfortable using in preparing a streaming program. ITS is investigating transcoding equipment that would allow us to record/store in one format, but deliver multiple formats in real-time.

The following is a quick explanation of some of the basic terms you will encounter in streaming media discussions.

CO-DEC - a COmpression-DECompression algorithm used for efficient storage and transmission of a data file, such as a video or audio file.

MPEG - the Moving Picture Experts Group defines industry standards for the encoding, management, and delivery of content through various media.

MPEG-1 - a codec designed for near-VHS quality video. MPEG-1 is primarily designed for CD-ROM delivery of content through various media.

MPEG-2 - a codec designed for high-quality video. MPEG-2 is primarily used for DVD disc encoding and other high-quality archival solutions, but can be streamed over high-bandwidth connections, such as Internet2. MPEG-2 playback often requires additional software and/or hardware.

MP3 - a codec designed for audio. MP3 is the most popular standard used for distribution on the Internet and in portable music players, such as Apple’s iPod. Note that MP3 stands for MPEG Audio Layer 3, not MPEG-3 (there is no MPEG-3).

MPEG-4, QuickTime, Real, and Windows Media are all streaming media architectures. They can handle a variety of media such as audio, video, text, animation, and 3D. They also support some forms of metadata (information about the media asset) and interactivity.

MPEG-4 - built around non-proprietary codecs, MPEG-4 excels at multi-architecture compatibility. Future codecs should enhance quality.

QuickTime - built around several non-proprietary codecs, QuickTime excels at medium to high bandwidth clips. It supports MPEG-4.

Real - built around proprietary RealNetworks codecs, Real format excels at low to medium bandwidth clips. It supports MPEG-4. Producers can use the SMIL language to add interactivity.

Windows Media - built around proprietary Microsoft codecs, Windows Media excels at medium bandwidth clips. It does not support MPEG-4.

Cisco IP Telephony Solutions Pass Deployment Tests for U.S. Department of Defense

Required JITC Tests Confirm Cisco Solutions' Interoperability, Reliability, Resilience and Security
RESEARCH TRIANGLE PARK, NC., September 27, 2004 - Cisco Systems®, Inc. today announced that it has passed U.S. Department of Defense (DoD) Voice over Internet Protocol (VoIP) interoperability tests. Certification by the DoD's Joint Interoperability Test Command (JITC) confirms that all aspects of Cisco's IP telephony solutions conform to the interoperability, reliability, resilience and security requirements of the DoD's multi-vendor voice network.

Cisco has achieved JITC PBX2 certification, allowing DoD organizations to deploy Cisco IP Telephony solutions for switched call connections. Cisco is now working toward PBX1 certification, which would allow DoD organizations to migrate Command and Control users to the same Cisco infrastructure.

"This certification represents a significant milestone for enterprise IP networks," said Captain Chris Christopher, Deputy Director for Future Operations, Communications, and Business Initiatives of the U.S., Navy's NMCI Office, who is responsible for emerging technologies and solutions within the world's largest Intranet. "It indicates that voice over an IP network has reached a level of maturity in the sector lifecycle that allows it to be viewed as an application like any other application. The PBX2 certification is an exciting development along with the other VoIP certifications that have happened in recent months. NMCI sees VoIP as a horizontal IP application which is vendor agnostic, allowing for a variety of choices combined with simplifying enterprise management."

The Cisco IP telephony system enables the transmission of data, voice, and video traffic over a single Cisco AVVID (Architecture for Voice, Video and Integrated Data) infrastructure. DoD organizations deploying Cisco IP telephony stand to realize increased productivity, greater business flexibility and reduced operational costs that come from a converged network. The Cisco IP telephony system also establishes a solid foundation for the deployment of advanced, feature-rich services such as unified messaging, multimedia conferencing, collaborative contact centers and interactive multimedia response systems.

"Our AVVID infrastructure will ultimately enable DoD agencies to achieve their long-term vision of network-centric operations, enabling unprecedented collaboration capabilities and speed of communications," said Bruce Klein, Federal area vice president, Cisco Systems. "Deploying our JITC-certified IP telephony solution is a good incremental step in that process."

Added Ed Carney, vice president of Cisco's Government Systems Unit, "JITC certification underscores the results of other independent tests that demonstrate the high level of security and stability of Cisco IP telephony solutions in even the most difficult environments. It also assures DoD customers that our solution will interoperate with their existing equipment and enable them to migrate to IP at their own pace."

The certified elements include:

Cisco CallManager 3.3 call processing software, extending enterprise telephony features and capabilities to packet telephony network devices such as IP phones, media processing devices, voice over IP gateways and multimedia applications;
Cisco Catalyst 3550 switches, stackable, multilayer switches that provide high-performance IP routing, advanced quality of service (QoS), and enhanced data security across the network, while maintaining the simplicity of traditional local area network (LAN) switching;
Cisco Catalyst 4500 and 6500 switches, supporting converged services from the wiring closet to the core to the wide area network (WAN) edge, and
Cisco 2600 and 3700 gateways, providing on-board LAN/WAN connectivity and new high-density service modules in a compact form factor.

SIP Architecture

IP COMMUNICATIONS AND SIP

Converged IP network technology is a reality. Cisco has been delivering these types of productivity benefits for years, with solutions built on Cisco AVVID (Architecture for Voice, Video and Integrated Data), a blueprint for building secure, high-performance converged IP networks.
SIP is a peer-to-peer, multimedia signaling protocol that integrates with other Internet services, such as e-mail, Web, voice mail, instant messaging, multiparty conferencing, and multimedia collaboration. When used with an IP infrastructure, SIP helps to enable rich communications with numerous multivendor devices and media. SIP can set up individual voice or conference calls, videoconferences and point-to-point video-enabled calls, Web collaboration and chat sessions, or instant messaging sessions between any number of SIP enabled endpoints, including IP phones, PCs, laptops, personal digital assistants (PDAs), and cell phones. In the opening scenario, the participants could be using end devices from any number of different vendors, and if the devices supported the necessary SIP applications with sufficient attention paid to implementation, the rich-media conference call would work perfectly.
SIP is an IETF standard that promises to open up IP communications networks to new hardware and software players, giving enterprises more options and flexibility in building converged networks. At one time, enterprises that employed time-division multiplexing (TDM)-based PBXs had to rely on the PBX vendor to supply any required features and functions; now, converged IP networks and SIP open up the application development process, allowing applications from independent software vendors with expertise in specific vertical markets. This process is enabled by the approach the IETF has taken to SIP which is defining the base-level functions required for interoperability, but leaving room for differentiation at the application level.
Cisco has been instrumental in defining SIP standards. The company has been at the forefront of SIP technology since the first SIP IETF RFC was published in 1999. Cisco engineers currently co-chair both the SIP and the related SIPPING working groups, and the company has been delivering SIP-enabled products since 2000. Cisco has participated in numerous multivendor SIP interoperability and test events, and is a founding member of the SIP Forum industry group. In delivering SIP-based solutions, Cisco draws on years of experience building converged IP networks for enterprises as well as service providers-an advantage that is unique in the industry

HelloGateway

VoIP gateways provide an interface between analog POTS phones, legacy PBXs, FAX machines, Public Switched Telephone Networks and packet switched networks. These help in protecting the existing investment in analog phones and fax machines, while taking advantage of the benefits of IP telephony. Additionally, VoIP gateways provide the fastest, simplest and easiest mechanism to leverage the benefits of IP Telephony on traditional PSTN infrastructure.

HelloSoft provides a comprehensive software suite for Signaling and Voice Media Processing. The solution includes SIP Signaling Stack, Voice Codecs, Line Echo Canceller, Jitter Buffer, Media and System Frameworks, Call Manager and all necessary Software components required for VoIP Gateways. The solution is standards compliant and has been tested for conformance and interoperability with commercial SIP proxy servers and popular IP Phones available in the market. Using this solution, Semiconductor and ODM/OEM customers can deliver ATA and VoIP Gateway products with a very short time to market schedule.

HelloSoft’s VoIP software stack is the most optimized software suite in the industry, from both a MIPS utilization perspective as well as a memory foot-print perspective. The entire solution is targeted on a single RISC processor making it extremely attractive for lowest cost ATA and Gateway designs.

The software is standards compliant solution for building either FXS-only Analog Telephone Adapters or, FXS/FXO gateways and FoIP gateways targeted towards SOHO and Residential markets, it has been proven on different hardware platforms and operating systems and tested for interoperability with different SIP based PSTN gateways and FoIP gateways.

Salient features:
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Comprehensive solution for FXS/FXO/FoIP gateways that can be easily tailored to reduce memory foot print

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Support for up to 8 voice channels

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Designed for a single RISC processor-based solutions

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Low foot print – memory and processing resources

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Available for popular Operating Systems like Linux/MVLinux, eCos, ThreadX, WinCE/PocketPC and others

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Software is Operating System Agnostic, provides OS abstraction layer for porting to new OS in very short time

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Available for ARM, MIPS, OMAP architectures

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Well defined Application Layer API’s facilitate easy integration with custom applications

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ITU-T G.168 compliant Line echo canceller

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All components and stacks are standards compliant

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Field proven and licensed by various major semiconductor manufacturers and ODMs/OEMs

HelloIP-Phones

IP Phones work on the principle of using the IP/Data networks for signaling and voice media transport. These phones provide a rich set of features compared to those available with traditional PSTN phones. IP Phones can be used with wire-line and wireless network interfaces for accessing packet networks and are meant for enterprise as well as residential applications.

HelloSoft provides the most optimized and comprehensive IP Phone software suite for Signaling and Voice Media Processing. The solution includes a comprehensive SIP Stack, Voice Codecs, Acoustic Echo Canceller, Jitter Buffer, Media and System Frameworks, Call Manager and all necessary software components required to build an IP Phone. It has been proven on different hardware platforms and operating systems. The solution is standards compliant and has been tested for interoperability with commercial SIP proxy servers and popular IP phones available in the market. Using this solution, Semiconductor and ODM/OEM customers can deliver IP Phone products with a very short time to market schedule.

HelloSoft VoIP software suite is the most optimized software suite in the industry, from both a MIPS utilization perspective as well as a memory foot-print perspective. The entire solution is targeted on a single RISC processor making it extremely attractive for lowest cost IP Phone designs.

Salient features:
- Targeted for wireline and wireless IP Phones (VoWLAN Handsets)
- Standards compliant, Complete Software solution for IP Phones, based on single RISC processors
- Industry’s lowest foot print – memory and processing resources
- Available for popular Operating Systems like Linux / MVLinux, eCos, ThreadX, WinCE / Pocket PC and others

- Software is Operating System Agnostic, provides OS abstraction layer for porting to new OS in very short time

- Available for ARM, MIPS, OMAP architectures
- Well defined Application layer API’s facilitate easy integration with the custom applications
- High quality Acoustic Echo Cancellation for Full-duplex Speaker Phone
- IMS capable design
- Field proven and licensed by major semiconductor manufacturers and ODMs/OEMs

How VoIP / Internet Voice Works

VoIP services convert your voice into a digital signal that travels over the Internet. If you are calling a regular phone number, the signal is converted to a regular telephone signal before it reaches the destination. VoIP can allow you to make a call directly from a computer, a special VoIP phone, or a traditional phone connected to a special adapter. In addition, wireless "hot spots" in locations such as airports, parks, and cafes allow you to connect to the Internet and may enable you to use VoIP service wirelessly.