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August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Quality and Network Performance Mike Moldovan Director of Engineering, Telephony, Slide 2 August 3-4, 2004 San Jose, CA www.voipdeveloper.com Agenda Telecom World VoIP Challenges VoIP Testing Solutions Summary Slide 3 August 3-4, 2004 San Jose, CA www.voipdeveloper.com Telecom World Slide 4 August 3-4, 2004 San Jose, CA www.voipdeveloper.com Telecom World Todays 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 Slide 5 August 3-4, 2004 San Jose, CA www.voipdeveloper.com Telecom World Todays 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 Slide 6 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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) Slide 7 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 8 August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Challenges Slide 9 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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% Slide 10 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 11 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 12 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 13 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 14 August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Test Solutions Slide 15 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 16 August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Test Solutions Main Interest Areas 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 3 rd party protocol implementations Slide 17 August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Test Solutions Main Interest Areas 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 Slide 18 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Bulk Call Generation for Enterprise Slide 19 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 20 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 21 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 22 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 23 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 24 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 25 August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Test Solutions IP Telephony Migration Full Set of Testing Solutions Pilot During deployment Post deployment Slide 26 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 27 August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Test Solutions Quality of Fax 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 fax traffic that is further analyzed Slide 28 August 3-4, 2004 San Jose, CA www.voipdeveloper.com VoIP Test Solutions Load 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 fax traffic that is further analyzed Slide 29 August 3-4, 2004 San Jose, CA www.voipdeveloper.com Summary Slide 30 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 31 August 3-4, 2004 San Jose, CA www.voipdeveloper.com 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 Slide 32 August 3-4, 2004 San Jose, CA www.voipdeveloper.com Questions?