2004 Conference Proceedings

Go to previous article 
Go to next article 
Return to 2004 Table of Contents 


FACT OR FICTION - TTY WORKS OVER VOICE OVER IP

Presenters
Don Pitchford
Cisco Systems
170 W. Tasman Dr.
San Jose, CA 95134
Phone: 408.902.3541
Email: dpitchfo@cisco.com

Overview

Reliable VoIP networks are a reality. There are over 10,000 enterprise VoIP deployments worldwide today with including half of the Fortune 500. More than 10 percent of the world's international long distance calls travel over IP networks. In fact, IP is establishing itself as the new standard for transport of voice and data communication and the basis for advanced telephony features in the future. VoIP is a serious alternative to proprietary circuit-switch phone systems and Cisco and Cisco customers are using TTY applications reliably today on Cisco VoIP networks.

This document:

Independent test results and real customers' TTY deployments on Cisco VoIP networks prove that packet loss is not a problem in properly engineered and implemented IP-based voice networks.

Facts about TTY and VoIP

TTY traffic can be successfully transmitted over an IP network with less than 1% Total Character Error Rate (TCER), if the network is engineered properly. A properly engineered network is based on having end points that have the capabilities to proper encode the TTY into VoIP, as well as being capable of delivering time-sensitive packets accurately and timely.

For TTY to be successfully transmitted over a data network, it must use a non-compression encoder / decoder (CODEC), such as G.711. Using compression CODEC, such as G.729a, will distort the TTY tones causing errors and makes the TTY conversation un-readable. Besides using the appropriate CODEC, the network should also enable quality of service (QoS). QoS prioritize voice & TTY packets are delivered. QoS also helps prioritizing the Voice & TTY traffic over data traffic, thus guaranteeing consistent quality of service. The effect of packet loss on TTY is great. Having a network with more than 0.12 percent packet loss can result in greater than one percent Total Character Error Rate (TCER), which will make the network not compliant to Section 508 or usable by TTY machines.

On top of all of this, the end point has significant role ensuring QoS. If you are connecting via acoustic coupling, the handset may not seat properly in the TTY device, or the phone itself may not be able to prioritize the voice/TTY traffic over the data coming from the PC. This in fact could result in poor quality of service. For direct connect, which is preferred method because the TTY is not subject to environmental noise, the device should support the necessary QoS methodology to ensure that the TTY traffic is given same priority as voice traffic.

VoIP is capable of supporting TTY traffic, however, consideration must be made to the network to ensure support of both voice and TTY. These same considerations are what are being used to design most enterprise voice networks today.

Packet Loss

Packet loss refers to the packets of data that are dropped by a network to manage congestion. Data applications are very tolerant to packet loss, as they are generally not time sensitive and can retransmit the packets that were dropped.

Applications such as speech or TTY traffic are time sensitive and cannot afford to wait for a dropped packet to be re-transmitted. Dropped packets in a VoIP network appear as noise or gaps in the conversation and may require the speaker to repeat their last word or sentence. For speech, there are methods for concealing lost packets. The human brain does a wonderful job of piecing the information together. However with TTY, a dropped packet may mean a garbled message and there is no recovery for the dropped packet. By properly engineering and implementing VoIP, the network will be able to differentiate between data packets and voice packets and manage the congestion to mitigate lost packets for time-sensitive data, such as voice and TTY traffic.

Cisco QoS - Enables Reliable Voice & TTY over IP

Quality of Service (QoS) refers to the capability of a network to provide better service to important network traffic. In the case of VoIP, this means providing reliable delivery of time sensitive content such as voice and TTY traffic. Introduced in Cisco product lines as early as 1995, QoS is now generally available on IP equipment from nearly all manufactures.

The fundamental of QoS is to manage bandwidth, delay, jitter and packet loss. By deploying several mechanisms, such as providing precedence to Voice or TTY traffic over non real-time data, QoS can be guaranteed deliver of the important voice or TTY traffic. Other methods include echo cancellation, dynamic jitter buffer, and packet concealment for loss packets. For additional information on QoS, please see: http://www.cisco.com/networkers/nw01/pres/preso/IPServiceandTechnologies/IPS-130.pdf

Dispelling the "Packet Loss" Myth

While a limited number of traditional PBX vendors claim that enterprise VoIP networks are incapable of supporting TTY devices, due to "inevitable packet loss" and "failure of VoIP to deliver packet loss, rates are less than 0.12 percent to 0.5 percent. Cisco and Cisco customers reliably operate TTY devices on Cisco VoIP networks. These facts speak for themselves.

Packet Loss IS NOT inevitable

Miercom (www.miercom.com), a network consultancy group specializing in network and communication related product testing and analysis, evaluated the QoS performance of the Cisco 7206 router in 1999. Miercom's testing involved deliberate, sustained overloads of an IP route through a T1 line, and an analysis of what happened with QoS turned off, versus QoS when turned on. Miercom found that there was "no loss at all of the high-priority traffic with prioritization compared to heavy loss without it"

Cisco's load testing of IP networks, based on Cisco equipment with QoS implemented, shows zero percent IP voice packet loss, even under severe network "overload" conditions.

Public Carrier IP Backbones deliver Packet Loss less than 0.12%

vAs an example of how public carriers handle IP packet loss rates, AT&T's U.S. IP Backbone number for July of 2002 shows that total IP packet loss was 0.02 percent . This is 25times better than the myth of a 0.5 percent rate and six times better than a 0.12 percent - a rate deemed unachievable by some PBX suppliers.

Other major IP carriers show similarly excellent performance for their IP backbones. AT&T's real-time IP backbone statistics can be found on this web site. http://ipnetwork.bgtmo.ip.att.net/index.html

In addition, Cooperation Association for Internet Data Analysis (CAIDA), provides tools and analyses promoting the engineering and maintenance of robust, scalable global internet infrastructures. Their web page has links to major IP backbone service providers' web pages, where they track the performance of their respective IP networks. www.caida.org

Conclusion

A Cisco VoIP/IP telephony system can be set-up to provide reliable G.711 to and from any Cisco IP phone designated on a Cisco IP PBX. This can provide the TTY end-user with G.711 at their endpoint. Secondly, Cisco IP phones have a built-in switch that can prioritize voice/TTY traffic over data traffic from the PC. Also, Cisco is the leading expert in designing QoS networks. Cisco is also working with international standards bodies to try to develop standards for transmission of TTY across any CODEC successfully.


Go to previous article 
Go to next article 
Return to 2004 Table of Contents 
Return to Table of Proceedings


Reprinted with author(s) permission. Author(s) retain copyright.