August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com VoIP Quality and Network Performance Mike Moldovan Director of Engineering, Telephony,
Jan 01, 2016
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
VoIP Quality and Network Performance
Mike Moldovan
Director of Engineering, Telephony,
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
Agenda
• Telecom World
• VoIP Challenges
• VoIP Testing Solutions
• Summary
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
Telecom World
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
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
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
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
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)
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
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
VoIP Challenges
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%
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
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
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
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
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
VoIP Test Solutions
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
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 3rd party protocol implementations
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
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
Bulk Call Generation for Enterprise
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
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
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
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
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
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
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
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
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
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
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
Summary
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
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
August 3-4, 2004 • San Jose, CA • www.voipdeveloper.com
Questions?