Ixload - Voice Test Solution: Validate That Your Voice Service Delivers A High Quality of Customer Experience
Ixload - Voice Test Solution: Validate That Your Voice Service Delivers A High Quality of Customer Experience
SOLUTION
KEY FEATURES:
• Emulate real-world traffic using Ixia’s highly scalable test platform or virtual ports
• Simulate detailed call control state machines, messages, and contents and create any test case,
including negative testing
• Integrate with the real-time transport protocol (RTP) test library to generate voice, conversational
video, and tones, and use a multitude of voice and video codecs to measure voice or video quality
• Fully automate feature and regression testing using the IxLoad TCL, REST, Python or Perl API
PRODUCT CAPABILITIES
IxLoad in a typical configuration simulating SIP endpoints and SIP proxy to test a distributed application
layer gateway (ALG)
IxLoad provides IMS system and sub-system developers and providers with a complete solution to verify
all aspect of IMS systems and their components. IxLoad is an invaluable tool for service providers and
network equipment manufacturers (NEMs) for functional, protocol, conformance, regression and
performance testing of end-to-end IMS networks and individual components. A single test can be used
for both functional and load testing, on a single hardware platform.
IxLoad in a typical configuration simulating SIP user endpoints, MMEs, and eNodeBs
for end-to-end LTE / IMS testing
SCENARIO EDITOR
The variety of implementations and protocol interpretations of VoIP, UC, IMS, and VoLTE solutions poses
interoperability challenges not just for networks, but for testing as well. Test tools need to accommodate
the variations and provide high flexibility in terms of the call flow (sequence of messages). IxLoad’s
Scenario Editor provides a powerful yet intuitive tool to edit the call flow and the content of control
messages. It allows test engineers to address any interoperability issue and emulate any supplemental
service for functional and negative testing. Innovative features allow users to easily create and modify test
scenarios using a complete visual environment in which the technology-specific test functions and
predefined test procedures are represented as visual blocks. Without any prior knowledge of
programming or script languages, users can easily define tests by simple drag-and-drop operations.
Scenario Editor: a test scenario for emualting SIP endpoints originating and terminating calls is
built by connecting functional blocks
The same test scenario can be used, without changes, at low scale or to emulate thousands or millions of
endpoints or subscribers. This is achieved by the parameterization capabilities—the use of variables,
expressions, and control functions—provided by the Scenario Editor.
EVENT VIEWER
IxLoad provides a large set of statistics across metrics of interest for call control and media. However, the
stochastic analysis of voice QoE it is not always sufficient. To identify and debug issues, it is not enough
to know the percentage of the calls or subscribers with issues, it is important to know which call or
subscriber failed. IxLoad’s Event Viewer contains a great capability to pinpoint the exceptions and errors.
Event Viewer showing the errors for each emulated user endpoint
During a test, Event Viewer displays and logs error events per emulated endpoint, facilitating the
diagnosis and correction of issues. By getting this information right away, test engineers can focus on
addressing the issue instead spending time analyzing massive traffic captures.
SPECIFICATIONS
SIP
Emulation SIP endpoints and user agents, SIP endpoints behind SIP proxies, SIP
servers (proxy, registrar), IMS entities, trunks, gateways
Features • Maintains full control over SIP state machines, messages, and contents;
creates any functional and load test case, including negative testing
• Drag-and-drop GUI permits functional building blocks to be easily
assembled into test cases and call flows with automatic protocol rule
enforcement
• Session timers and message retransmission mechanisms are present
with the option to be disabled by the user
• When in SIP Proxy mode, the module routes the SIP messages based on
content of the SIP messages, following the user defined rules
• When in SIP Proxy mode, the library has the capability to interact with an
HSS emulating the Cx interface
• Supports WebRTC websockets
• Capability to test IMS devices and networks; x-CSCF isolation testing and
emulation
SIP
• Includes IMS call profiles
• Graceful stop at ramp-down to end all the active calls at the end of test
• Test cases built for functional and feature testing can be reused for stress
testing
• Emulates real-world traffic using Ixia’s highly scalable test platform
• Simultaneously supports data, voice, and video protocols to emulate
multiplay subscriber environments
• Tests a device’s ability to sustain designed load levels or call attempt /
connection rate
• Supports custom load profiles, which contain individual settings for each
call mix element
• Performs call feature interoperability testing
• Provides ladder diagrams and media decoding with built-in packet
capture and analyzer for in-depth SIP and RTP stream analysis
• Ships with library of pre-built test cases and call flows for easier startup
Network Capabilities • Link layer protocols, including PPPoE, IPsec, and DHCP
• Transport layers: UDP, TCP, TLS, WebSocket, Secure WebSocket
• Diffserv type of service (TOS/DSCP)
• Real-world network configurations: multiple sub-networks, unique MAC
addresses, 802.1q, 802.1p, and emulated router support
• VLAN tag with Q-in-Q support
• Configurable MAC addresses
SIP
• SIP Call Times, SIP Delays, SIP Registrations,
• SIP Registration Rates, SIP Messages, SIP Errors
• VoIP/SIP Errors, SIP Busy Hour Call, SIP Cloud, SIP Other
Media Capabilities • Integrated with the RTP test library to generate voice, DTMF, tones, and
video; supports a multitude of voice codecs and the ability to test voice
quality
• Supports Video Conference and Telepresence
• Emulates SIP endpoints submitting / receiving SMS
• Negotiates sessions with Fax over IP (T.38)
• Message Session Relay Protocol (MSRP)
VOLTE
Protocols Supports stateful emulation of SIP and other Layer 7 protocols over eGTP
to provide a comprehensive solution for VoLTE testing:
Control plane
• 3GPP TS 29.274 Evolved General Packet Radio Service (GPRS)
Tunneling Protocol for Control plane (GTPv2-C)
• 3GPP TS 29.281 GPRS Tunneling Protocol User Plane (GTPv1-U)
• IPv4 transport
VOLTE
• QPSK, 16QAM and 64QAM modulation schemes
• NAS compression and ciphering
• Full DL/UL HARQ capability
• Semi-persistent scheduling
• UE power control, group hopping
• Frequency hopping modes 0, 2, 4
• Automatic configuration of MIB/SIB parameters
• Support for default and dedicated bearers, with a maximum of 11 per UE
• UE initiated dedicated bearer creation, modification and deletion
• Network initiated dedicated bearer creation, modification
• Dynamic IP address allocation during session establishment
• Supports all LTE handover types
• Configuration of QoS and TFT per layer 7 activity
GTP-c Features • Support for default and dedicated bearers, with a maximum of 11 per UE
• UE/MME initiated dedicated bearer creation, modification and deletion
• Network initiated dedicated bearer creation, modification
• Echo request and response
• Dynamic IP address allocation during session establishment
• Ability to configure the number of simulated MMEs and eNodeBs
• Supports eNodeB (X2) handovers with configurable event intervals and
mobility paths between eNodeBs
• Supports S1-based handovers with indirect data forwarding tunnels
• Configuration of IMSI, MSISDN, IMEISV, RAC/LAC/TAC, MCC/MNC
• Configuration of QoS and TFT per layer 7 activity
• Prebuilt TFTs are supplied, with the ability to manually override with user
TFT definition
WEBRTC
WEBRTC
WEBRTC
Commands Make Call, Accept Call, End Call, Reject Call, Initiate Modify, Receive Modify,
and Media Session
Media Capabilities • Integrated with the RTP test library to generate voice, DTMF, tones and
video. Supports a multitude of voice codecs and the ability to test voice
quality.
• Supports Video Telephony and Cisco Telepresence
• DTLS support for SRTP key negotiation
H.248 (MEGACO)
Transport UDP
Features • Maintains full control over H.248 state machines, messages, and contents.
Allows the creation of any test case, including negative testing.
• Drag and drop GUI permits functional building blocks to be easily assembled
into test cases and call flows with automatic protocol rule enforcement
• Tests a device’s ability to sustain designed load levels
• Supports custom load profiles, which contain individual settings for each call
mix element
• Supports call feature testing under load
• Performs call feature interoperability testing
• Provides ladder diagrams and media decoding with built-in packet capture and
analyzer for in-depth H.248 and RTP stream analysis
• Ships with library of pre-built test cases and call flows for easier startup
H.248 (MEGACO)
Network Capabilities • Link layer protocols, including PPPoE, IPsec, and DHCP
• Diffserv type of service (TOS/DSCP)
• Real-world network configurations: multiple sub-networks, unique MAC
addresses, 802.1q, 802.1p, and emulated router support
• VLAN tag with Q-in-Q support
• Configurable MAC addresses
H.248 Profiles • "ETSI_ARGW" – ETSI H.248 Profile for controlling Access and
Residential Gateways (NGN Release 1) / Reference doc: ETSI ES 283
002 V1.1.3 (2007-07)
• "ETSI_GateControl" – ETSI H.248 Profile for controlling gates between IP
transport domains / Reference doc: ETSI TS 102 333 V1.2.0 (2008-01)
• "ETSI_BGF" – ETSI H.248 Profile for controlling Border Gateway
Functions (BGF) in the Resource and Admission Control Subsystem
(RACS) v1 / Reference doc: MSF-IA-MEGACO.009-FINAL
• "ETSI_BGF" – ETSI H.248 Profile for controlling Border Gateway
Functions (BGF) in the Resource and Admission Control Subsystem
(RACS) v2 / Reference doc: ETSI ES 283 018 V2.3.0 (2008-05)
• "ETSI_TGW"– ETSI H.248 Profile for controlling Trunking Media
Gateways / Reference doc: ETSI ES 283 024 V1.1.4 (2007-07)
• "MSF TGW" – MSF H.248 Profile for controlling an IP Trunking Gateway
(Implementation Agreement Between a Call Agent and an IP Trunking
Gateway) / Reference doc: MSF-IA-MEGACO.003.01-FINAL
• "MSFR3TGW” – MSF H.248 Profile for controlling Trunking Media
Gateways (Implementation Agreement for a MSFR3 MGC-2 Interface) /
Reference doc: MSF-IA-MEGACO.011-FINAL
• "MSF_BGF" – MSF H.248 Profile for Distributed Session Border
Gateways (S-SBG/P-CSC to D-SBG Interface Implementation
Agreement) / Reference doc: ETSI ES 283 018 V1.1.4 (2007-10)
• "MSFUKAG" – MSF H.248 Profile for multi-service Access Gateway /
Reference doc: MSF-IA-MEGACO.005-FINAL
H.248 (MEGACO)
• ETSI_BGF/1
• ETSI_BGF /2
• MSF_BGF/1
• MSFUKAG/1
Media Capabilities • Integrated with the RTP test library to generate voice, DTMF, tones, and
video. Supports a multitude of voice codecs and the ability to test voice
and video quality
MGCP
Features • Maintains full control over MGCP state machines, messages, and
contents. Allows the creation of any test case, including negative testing.
• Permits easy assembly of functional building blocks into test cases and call
flows with automatic protocol rule enforcement through a drag and drop GUI
• Tests a device’s ability to sustain designed load levels
• Supports custom load profiles, which contain individual settings for each
call mix element
• Supports call feature testing under load
• Performs call feature interoperability testing
• Ships with library of pre-built test cases and call flows for easier startup
MGCP
Network Capabilities • Link layer protocols, including PPPoE, IPsec, and DHCP
• Diffserv type of service (TOS/DSCP)
• Real-world network configurations: multiple sub-networks, unique MAC
addresses, 802.1q, 802.1p, and emulated router support
• VLAN tag with Q-in-Q support
• Configurable MAC addresses
Commands • Send CRCX, Send DLCX (CA), Send MDCX, Send RQNT
Simulated CA
• Send AUEP, Send AUCX, Send EPCF, Wait NTFY
• Wait DLCX (CA), Wait Command (CA), Wait RSIP
Commands • Send NTFY, Send DLCX (CA), Send RSIP, Wait CRCX
Simulated MGW
• Wait DLCX (GW), Wait MDCX, Wait RQNT, Wait AUEP
• Wait AUCX, Wait EPCF, Wait Command (GW)
Media Capabilities • Integrated with the RTP test library to generate voice, DTMF, tones, and
video. Supports a multitude of voice codecs and the ability to test voice
and video quality
SCCP (SKINNY)
SCCP (SKINNY)
• Tests the call manager’s ability to sustain designed load levels
• Supports custom load profiles, which contain individual settings for each
call mix element
• Supports call feature testing under load
• Performs call feature interoperability testing
• Simulates device stations: 7902, 7910, 7935, 7940, 7960, 30 SP+,
12SP +, 12 SP, 12 S, 30 VIP, VGC
• Ships with library of pre-built test cases and call flows for easier startup
Network Capabilities • Link layer protocols, including PPPoE, IPsec, and DHCP
• Diffserv type of service (TOS/DSCP)
• Real-world network configurations: multiple sub-networks, unique MAC
addresses, 802.1q, 802.1p, and emulated router support
• VLAN tag with Q-in-Q support
• Configurable MAC addresses
Media Capabilities • Integrated with the RTP test library to generate voice, DTMF and tones.
Supports a multitude of voice codecs and the ability to test voice quality.
Features • The RTP/SRTP engine and the voice and tones functions are integrated
with all VoIP signaling modules
• media capabilities are negotiated during the call setup phase using the
automated generated or user defined SDP
• Maintains control of SDP state and generates media accordingly the SDP
negotiated parameters
• Performs real time quality of service measurement using various metrics:
o MOS R-Factor for audio,
o PESQ Perceptual Evaluation of Speech Quality
o POLQA Perceptual Objective Listening Quality Assessment
• Plays real media clips and any user defined sequence of tones
• Detects and validate tone sequences
• Supports multiple type of media in same call and same test
Network Capabilities • Link layer protocols including: PPPoE, IPsec, DHCP, GTP
• Diffserv type of service (TOS/DSCP) for each type of traffic
• Real-world network configurations: multiple sub-networks
o unique MAC addresses
o 802.1q
o 802.1p
o and emulated router support
• VLAN tag with Q-in-Q support
• Configurable MAC addresses
Tones • Real time Generation and Detection of DTMF (Dual Tone Multiple
Frequency), MF (Multi-Frequency tones) or Single Tones sequences
• User configurable sequences
• User configurable MF and Single tones
• In band and out of band tone transmission
• 2833 Events and 2833 Tones
Features • The RTP/SRTP engine and the conversational video functions are
integrated with SIP - including VoLTE and WebRTC - H.248, MGCP and
Skinny libraries;
• media capabilities are negotiated during the call setup phase using the
automated generated or user defined SDP
• Maintains control of SDP state and generates media accordingly the SDP
negotiated parameters
• Performs real time quality of video (VQMon)
• Plays real media clips from an open pool of video clips
• Supports Cisco Telepresencetm endpoint emulation
• Support for Multiple stream per endpoint
• Speaker rotation scheme for video conferencing testing:
o All speak, Sequential, Random
• Supports multiple type of media in same call and same test
Network Capabilities • Link layer protocols including: PPPoE, IPsec, DHCP, GTP
• Diffserv type of service (TOS/DSCP) for each type of traffic
• Real-world network configurations: multiple sub-networks
o unique MAC addresses
o 802.1q
o 802.1p
o and emulated router support
• VLAN tag with Q-in-Q support
• Configurable MAC addresses
PLATFORM OPTIONS
VISIT IXIACOM.COM FOR MORE INFORMATION ON
IXLOAD PLATFORM OPTIONS
Note: Some of the IxLoad Voice features are available only on specific load modules
TECHNOLOGY SOLUTIONS
VISIT IXIACOM.COM FOR MORE INFORMATION ON
IXLOAD TECHNOLOGY SOLUTIONS
925-3372 IxLoad Voice over LTE-2016, Software Bundle, Layer 4-7 Performance Test
Application. Enables VoIP protocols for VoLTE testing. Includes: Advanced SIP &
RTP, Audio Codecs, WebRTC, IPsec and VoLTE extensions. Includes: Voice
Quality engine for up to 10Gbps, Video Quality engine for up to 10Gbps
conversational video traffic, AVDNET-DHCP to emulate DHCP enabled clients
and Software Impairment on selected hardware. Note: some of the features are
available only on specific load modules.
APPLIANCE LICENSES
PART NUMBER DESCRIPTION
925-6321 IxLoad Appliance Multiplay, Software Bundle, Layer 4-7 Performance Test
Application. Enables Data, Storage, Voice, Video and Access & VPN on
PerfectStorm ONE and Novus ONE appliances. Includes:
• 925-6121 IxLoad Appliance DATA & Storage
• 925-6112 IxLoad Appliance VIDEO
• 925-6113 IxLoad Appliance VOICE
• 925-6117 IxLoad Appliance ACCESS & VPN
925-6113 IxLoad Appliance VOICE, Software Bundle, Layer 4-7 Performance Test
Application. Enables IxLoad Voice functionality on PerfectStorm ONE and Novus
ONE appliances. Includes: Advanced SIP: SIP endpoint, Proxy and Cloud
emulation TLS, SRTP Audio, Video Conference, Telepresence, MSRP, 925-3528
Voice Quality engine for up to 10Gbps, 925-3512 Video Quality engine for up to
10Gbps conversational video traffic
939-9533 IxLoad VE Tier-3 10G Floating SUBSCRIPTION License. Includes the following
IxLoad protocols supported on IxLoad VE for a duration of 1-Year: Data (HTTP,
HTTPS, FTP, TFTP, DNS, DHCP, LDAP, Radius), Mail (IMAP, POP3, SMTP),
Storage (SMB, NFS, iSCSI, Storage I/O), Voice (VoIP SIP, VoLTE), Video
(DASH, Flash,HDS, HLS, IPTV VoD, MS IPTV, Silverlight), Ipsec, IxLoad- Attack
and IxLoad-AppLibrary. Enables 10 Gig throughput per unit.
939-9534 IxLoad VE Tier-4 10G Floating SUBSCRIPTION License. Includes the following
IxLoad protocols supported on IxLoad VE for a duration of 1-Year: Data (HTTP,
HTTPS, FTP, TFTP, DNS, DHCP, LDAP, Radius), Mail (IMAP, POP3, SMTP),
Storage (SMB, NFS, iSCSI, Storage I/O), Voice (VoIP SIP, VoLTE), Video
(DASH, Flash,HDS, HLS, IPTV VoD, MS IPTV, Silverlight), IPsec, IxLoad- Attack
and IxLoad-AppLibrary. Includes EPC and WiFi Offload protocols. Enables 10
Gig throughput per unit.