voip quality and network performance

August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com

VoIP Quality and Network Performance

Mike Moldovan

Director of Engineering, Telephony,


Agenda

• Telecom World

• VoIP Challenges

• VoIP Testing Solutions

• Summary


Telecom World


Telecom World – Today’s Changes

• The current telecom networks are based on circuit switching while the Internet is a packet switched network

• Circuit switching establishes a dedicated connection from end-to-end for the entire duration of the call

• Users have more complex and diversified needs• Telecoms and Enterprises migrate their infrastructure from PSTN

to IP• New multimedia services (video on demand, teleconferencing)

are provided on converged networks • Voice over IP:

– First application over IP that is truly real-time and requires the network to meet the demanding Quality of Service (QoS) performance.

– VoIP shall deliver the QoS that a normal telephone call offers today – No clipping of sounds, high intelligibility and no perceptible delay of user

speech in reaching the listener is a must


Telecom World – Today’s Changes

• Voice over IP (continued):– The large cost savings offered by Internet telephony is fueling rapid

growth in the global consumer market

– As consumers and telephone companies seek lower cost calling with a greater number of features, calling traffic will be shifted from the traditional telephone network to the Internet

– Because this shift is enabled through Softswitches and Media Gateways, they represent one of the most promising growth segments in the Internet telephony component market


Telecom World – VoIP Drivers

• Two main VoIP drivers:– Costs– New applications and Services

• Cost savings: less-expensive as compared with PSTN toll charges, lower bandwidth costs, reduced personnel

• More efficient network utilization

• Greater operational flexibility

• Integrated voice and data networks

• Convergence of voice, fax, data and video traffic

• New applications (i.e. video-conference, call centers, unified messaging)


Telecom World – VoIP Functional Layers

• VoIP Capable Networks– Internet, WAN, VPNs

• VoIP Enabled Infrastructure – Firewalls, Gateways, Routers

and Access Services

• IP Telephony – IP-PBX functionality, SIP and

directory services, Instant Messaging, IP Centrex

• Advanced IP Telephony Applications

– Contact Centers, Unified Messaging, Unified Communications, Conferencing, managed and outsourced IP-Telephony applications

Advanced IP Telephony

IP Telephony

VoIP Enabled Infrastructure

VoIP Capable Networks

Source: Gartner Research, 2003


VoIP Challenges


VoIP Challenges - QoS Obstacles

The biggest obstacle in the migration to VoIP networks is to assure customers receive the same QoS that they have come to expect in their current PSTN networks.

Customers expectations:– Pick up phone and get dial tone– Dial anybody and get connected every time– Power failures do not affect the telephone services– Telephone network availability is 99.999%


VoIP Challenges - QoS Obstacles• Network availability

– Connectivity issues– Early disconnects– Loss of speech path

• Voice– Latency – generates echo and talker overlap

– Echo – in VoIP networks delay is greater than 50 ms

– Jitter – de-jitter mechanisms induce more delay

– Packet loss – peak loads and congestions drops packets

• Fax and Data transmission– Timing – the fax precise timing can be skewed generating call loss– Packet loss – protocol can fail if information is lost or data exchange can

take more time (money) to be accomplished


VoIP Challenges - Testing Issues

• Complexity– Many types of IP telephony devices, especially during the transition

period from PSTN technology

– Multiple signaling protocols

– Multiple media types and CODECs

– Support for and quality of legacy analog services such as data modems and Fax must be assured

– Multiple traditional telephony call features and applications

• Scalability– Signaling– Media– Quality– Features


VoIP Challenges - Testing Issues

• Convergence– VoIP and PSTN must coexist for the foreseeable future

– The protocols developed for PSTN are migrated to VoIP generating new “hybrid” protocols

– Tests plans must include both VoIP and PSTN to be comprehensive

– Test equipment must account for both VoIP and PSTN technologies

– Legacy analog services, call feature testing and interfaces to external systems must be supported

– Assess Voice and Fax in the same manner is a must due to their critical nature for business communications


VoIP Challenges – Key Features in VoIP Testing

• Support for testing multiple VoIP protocols: SIP, H.323, MGCP, MEGACO, SIGTRAN, SCCP

• Support multi-interface PSTN to IP testing for assessing the telecom devices involved in the migration to IP

• Support all types of media streaming: Tones, Video, Voice, Fax, Data Modem

• Support open standards and test strategies that consider the interoperability with all different protocol implementations

• Ability to support multiple protocols and media streaming simultaneously for cross-technology testing (i.e. between Analog, TDM and VoIP interfaces).

• Ability to cross-analyze the test results in order to pinpoint the real DUT issues


VoIP Test Solutions


VoIP Test Solutions – Main Interest Areas

• Multi-endpoint simulation/ emulation – Simulation of hundreds or thousands VoIP and TDM end-

devices (i.e. IP Phones, Gateways, POTS).

– Implement basic (registration, call initiation and termination, hold) and advanced features (transfer, conference, IVR)

– Any combination of telephony interfaces testing

– Optimum breadth and depth of signaling protocols and media streaming (tones, voice, fax and data)

– Provide advanced QoV and QoF measurements



• Protocol Testing – Provide assessments of protocol functionality

– Access the low-level messages, load their structure from standard templates and modify them with valid or invalid values

– Use the low-level VoIP test functions for manipulating MGCP, SIP, H.323 and T.38 messages

– Verify the interoperability with 3rd party protocol implementations



• Bulk Call Generation for Enterprises – Generate hundreds or thousands of calls covering the area of

Enterprise Bulk Call Generation.

– Bulk call generation over broader coverage of VoIP signaling protocols (SIP, H.323, SKINNY) and media streaming (tones, voice, T.38 fax)

– Optimum solution for lower densities but need a mixture of standard and custom protocols


VoIP Test Solutions – Application Areas

Advanced IP Telephony

IP Telephony

VoIP Enabled Infrastructure

VoIP Capable Networks

Endpoint Simulation

Advanced Feature Testing

Bulk Call Generation for Enterprise

Endpoint Simulation

Basic Feature Testing

Protocol Testing




VoIP Test Solutions – Multi-endpoint simulation

• Call flow and feature testing • Call signaling• Media streaming• Quality of Service testing

– Latency, Jitter, Packet Loss, Packet Missordered– Quality of Voice– Quality of Fax

• Multi-interface testing (IP telephony migration)– IP to TDM, IP to Analog, Analog to TDM to IP, TDM to

IP to CTI


VoIP Test Solutions – Protocol Testing

• Generate and receive valid and invalid messages and flows – MGCP, SIP, H.323, SCCP, T.38

• Generate out-of-sequence messages– MGCP, SIP, H.323, SCCP, T.38

• Transactions that sends a command and wait for a specified response – MGCP, SIP, H.323, SCCP, T.38

• Quality of Service testing– Quality of Voice, Quality of Fax


VoIP Test Solutions – Bulk Call Generation

• Load testing– Generate hundreds, thousands of calls per system and calculate

BHCC

• Quality of Service testing under heavy load conditions – Latency, Jitter, Packet Loss, Packet Mis-ordered– Quality of Voice– Quality of Fax

• Multi-interface testing under load conditions– IP to TDM, IP to Analog, Analog to TDM to IP, TDM to IP to CTI


VoIP Test Solutions – Density vs. Features

• Optimum balance between density and number of telephony features covered

• Test telecom infrastructure and enterprise devices and equipments over their entire development lifecycle

Density to be Tested

Features to be Tested

Multi-interface,

Multi-protocol, Functionality

and Performance

Tester


VoIP Test Solutions – Call Flow

• Analog Remote Test Unit (RTU) originates the call

• VoIP RTU receives the call

• IxVoice verifies the flow of message between the System Under Test and the two RTUs


VoIP Test Solutions – IP-PBX

• The IP extension places a call to analog extension

• The analog extension transfers the call to the TAPI extension

• TAPI extension answers the call

• IP and TAPI extensions get connected


VoIP Test Solutions – IP Telephony Migration

Full Set of Testing Solutions• Pilot• During deployment• Post deployment


VoIP Test Solutions – Quality of Voice

• The first RTU places a call to the second one

• The IP traffic is altered using WAN Simulator

• The second RTU answers the call

• The two RTUs exchange voice traffic that is further analyzed


VoIP Test Solutions – Quality of Fax




• The two RTUs exchange fax traffic that is further analyzed


VoIP Test Solutions – Load




• The two RTUs exchange fax traffic that is further analyzed


Summary


Summary

• The VoIP Gateway market is projected to grow at a compounded annual growth rate of 229% and represents a market size of 1.8 billion

• The testing of VoIP devices is critical for manufacturers because these devices enable the explosive growth of Internet telephony

• Ixia IxVoice can cover the automated testing needs generated by the various VoIP device manufacturers


Summary

• Necessities of migration from PSTN to IP telephony networks at enterprise level has determined the device manufacturers to develop hybrid PSTN-VoIP solutions (such as hybrid IP-PBXs)

• Worldwide service providers are more and more interested in providing new and more reliable VoIP services

• Ixia IxVoice is specially designed to automatically test these devices and networks using a multi-interface, multi-technology approach


Questions?

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