SIP Protocol Emulator for Low- and High-Density Applications
Welcome to another issue of GL Communications newsletter providing information and insight into our MAPS™ SIP (Session Initiation Protocol) Emulator for low and high density applications from hundreds to tens of thousands of simultaneous calls, with signaling and traffic generation as per current standards.
SIP has become “the IP protocol” for voice connections (wired and wireless), video conferencing, instant messaging, and other forms of media, e.g FAX. Many signaling and traffic services are migrating to SIP, including emergency 911, push-to-talk for air traffic control, and group voice call services used by as First Responders.
GL's Message Automation & Protocol Simulation (MAPS™), a general protocol emulation architecture has been adapted for SIP testing, can simulate User Agents (User Agent Client- UAC, User Agent Server-UAS), Proxy, Redirect, and Registrar servers. As shown in the diagram, GL’s MAPS™ SIP test tool/traffic generator can be used to simulate any interface in a SIP network (standard SIP, SIP – I (ISUP), SIP IMS, and SIP MSRP), and perform SIP Protocol Conformance Testing (of SIP protocol implementations). SIP is also used in ED-137, the next generation air traffic control system.
A single MAPS™ SIP instance can act as more than one SIP entity and generate any SIP message in a VoIP network and hence equipment needed to test is reduced. MAPS™ framework allows running a set of tests/scripts sequentially, randomly, and simultaneously for multiple iterations. Users can schedule tests/scripts to run automatically on scheduled days and times. Bulk calls can be generated to simulate different load patterns.
The MAPS™ SIP Conformance Scripts is designed with 300+ test cases, as per the SIP specification of ETSI TS 102-027-2 v4.1.1 (2006-07) standard. Test cases include general messaging and call flow scenarios for multimedia to call session setup and control over IP networks. Test results are logged as pass/fail. Test cases verify conformance of actions such as registration, call control, proxies, and redirect servers.
GL’s MAPS™ SIP application is also available in the High-Density version (requires a special-purpose network appliance or portable platform), capable of high call intensity (hundreds of calls/sec) and a high volume of sustained calls (tens of thousands of simultaneous calls/platform).
- Unique endpoint emulation using IP address, MAC address, and VLAN tagging
- On a 2U grade server, high-density calls can sum up to 64,000 simultaneous calls with duplex RTP traffic
- Achieve up to 250 calls per second (with RTP traffic)
- Simulates around 50,000 to 100,000 endpoints
- Configurations, test scripts, and profiles can be saved and reused across different MAPS™ systems
- Real-time monitoring and reporting of Voice Quality, Signaling message counters, and call statistics
- Generates and processes SIP valid and invalid messages
- Supports IPv4 /IPv6 and transport over UDP and TCP, and TLS for secure transport
- Supports joining the conference call, unattended call transfer, attended call transfer, call hold, auto call rejection, and silence packets generation
- Ability to send "reliable provisional responses" and start early media actions
- Ability to implement IP Spoofing for any network like Class C, Class B, etc
- Supports in dialog and out of dialog transactions for SUBSCRIBE, NOTIFY, OPTIONS, REFER, and INFO SIP methods
- Supports simulating custom SIP messages and call scenarios
- Supports transmission and detection of various RTP traffic such as digits, voice file, single tone, dual tones, IVR, FAX, and Video in IP networks
- Supports almost all industry-standard codec types - G.711 (mu-Law and A-Law), G.722, G.729, G.726, GSM, AMR, AMR -WB, EVRC, EVS, OPUS, SMV, iLBC, SPEEX, and more. *AMR, EVRC, EVS, and OPUS variants require additional licenses
- Supports SRTP (Secure Real-time Transport Protocol)
- Provides Voice Quality statistics such as MOS, Packet loss, and Jitter
- Supports both RTP G.711 Pass-Through Fax (PKS200) and T.38 Fax(PKS211) simulation over IP
- Message Session Relay Protocol simulation supporting instant messaging
- Supports Interactive Voice Response (IVR) testing that recognizes and responds to voice prompts using DTMF digits or voice, allowing automated IVR traversal and testing
- Supports Short Message Service (SMS) over IP/IMS communication, SMS is encapsulated in a SIP message and carried over IMS core network
- API / CLI (Command Line Interface)
- MAPS™ CLI interface based on a client-server model allows users to control all features of MAPS™ through APIs
- Supported clients include Python, Java, TCL and others
- Fully integrated, complete test environment for SIP
- Supports end-to-end gateway testing
- Supports conformance testing UAC, UAS, Proxy, Registrars, Registrants, Redirect Servers, and other SIP entities
- Handles strict routing ™ lose routing, when requests are routed through proxies
- Functionality and load testing on SIP entities like Proxy, Registrar, IP PBX, B2BUA, VoIP Gateways, IP Phones, etc
- Testing NG9-1-1 emergency services (voice ™ text) and components within the ESInet