MAPS™ SIP Protocol Emulator
(Generate & Receive SIP Signalling with Traffic)
GL's Message Automation & Protocol Simulation (MAPS™) designed for SIP testing can simulate User Agents (User Agent Client- UAC, User Agent Server-UAS), Proxy, Redirect, Registrar and Registrant servers. This test tool/traffic generator can be used to simulate any interface in a SIP network and perform protocol conformance testing (SIP protocol implementations).
The application is available as
- MAPS™ SIP Protocol Test Tool (Item # PKS120)
- MAPS™ SIP Conformance Test Suite (Item # PKS121)
- MAPS™ HD Call Generator (Item # PKS109)
- MAPS™ SIP Message Session Relay Protocol (Item # PKS112)
The application gives the users the unlimited ability to edit SIP messages and control scenarios (message sequences). "Message sequences" are generated through scripts. "Messages" are created using message templates.
MAPS™ can be used to simulate any interface of the VoIP network. Single MAPS™ instance can act as more than one SIP entity at a time and can generate any SIP message on wire in VoIP network and hence equipment needed to test are reduced. MAPS™ SIP supports UDP, TCP, TLS transport types. Secure Real-time Transport Protocol (or SRTP) traffic supported over TLS (Transport Layer Security) Transport and TLS Library (OpenSSL) uses a Certificate and Key.
With the purchase of RTP Core license (PKS102), MAPS™ SIP supports transmission and detection of various RTP traffic such as, digits, voice file, single tone, dual tones, IVR, FAX*, and Video*. With regular RTP traffic, the maximum Simultaneous Calls up to 2500, and Calls per Second up to 250 is achievable. Almost all industry standard voice codec supported.
The MAPS™ SIP Conformance Scripts (PKS121) is designed with 400+ test cases, as per 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 call session setup and control over IP networks. Logging and pass/fail results are also reported. Test cases verify conformance of actions such as registration, call control, registrants, proxies and redirect servers.
MAPS™ SIP also supports generation of high volume of calls with traffic for load testing network using MAPS™ RTP HD (PKS109) network appliance.
MAPS™ RTP HD (requires a special purpose network appliance and PKS109 RTP HD licenses) is a specialized 1U rack mounted network appliance designed to easily achieve up to 20,000 endpoints per appliance (5000 simultaneous calls with duplex traffic per port). The appliance is available with 4x 1 Gbps NIC ports. Using a stack of multiple servers, a larger test system with 100K-200K calls (all controlled from a single Master Controller) is achievable for enterprise to carrier grade testing.
GL also provides a high-density network monitoring appliance (supports SIP, RTP, UMTS, LTE, and IMS) for online capture and decode of the signaling in real time.
** Some of these traffic types requires additional licenses – contact GL for more information
RTP traffic simulation is supported for almost all standard codecs. Also, supports various traffic events simulation during the course of a call, which is listed below:
- Answer Call - Used to Accept the Call from DUT
- Place Call - Places the Call to another End by initiating the Invite Message.
- Terminate - Terminates the call using BYE Method
RTP Traffic Events – digits, tones, files
- Send File, Receive File, Stop Send File
- Send Digits, Detect Digits, Stop Send Digits
- Send Test Tone, Detect Test Tone
- Send Tone, Detect Tone, Stop Send Tone
MAPS™ SIP provides the Bulk Video Call Simulation capability using pre-recorded video traces supporting codecs like H.264, H.263, and VP8. On a high-performance computing platform (core-i7), it is possible to generate more than 500 simultaneous video calls. It also provides statistics for the RTP traffic such as Listening MOS, Conversational MOS, PacketLoss, Discarded Packets, Out of Sequence Packets, Duplicate Packets, Delay and Jitter.
MAPS™ SIP supports FAX over IP (FoIP) simulation and monitoring. With Additional licensing, both RTP G.711 Pass Through Fax Simulation (PKS200) and T.38 Fax Simulation over UDPTL (PKS211) simulation are supported.
MAPS™ SIP supports Message Session Relay Protocol for instant messaging over SIP sessions (PKS112), simulating SIP/MSRP User Agents end-points in an NG9-1-1 network and send and receive communications over ESInets. The NG9-1-1 calls can be IM Only Calls, Audio and IM Calls, and Video and IM Calls between multiple UAs.
MAPS™ SIP handles simulation of Audio and Video along with IM call, during which the pre-recorded audio/video is sent and received using RTP and text messages using MSRP during the same call. During this call type, there will be three media lines, one for Audio, one for Video and another for text messages using MSRP.
MAPS™ SIP also supports command line interface to remotely control MAPS™ from client environment. Multiple MAPS™ CLI servers from the client environment (such as TCL, Python, VBScript, Java, or .Net) via APIs for complete automation.
GL also provides a Packet Analyzer for on-line capture and decode of the SIP signaling in real-time both during tests and as a stand-alone tracer for live systems.
- Generates and processes SIP valid and invalid messages
- Supports complete customization of SIP headers, call flow, and messages
- Supports complete customization of scripts and parameters in the profiles
- Each SIP message template facilitates customization of the protocol fields and access to the various protocol fields from the scripts
- Supports IPv4 /IPv6 and transport over UDP and TCP, and TLS for secure transport
- Handles Retransmissions of messages with specific interval
- Scripted call generation and call reception
- Supports 64-bit version to enhance signaling performance
- Supports conference call, blind call transfer, hold, auto call rejection, and silence packets generation
- Ability to send "reliable provisional responses" and start early media actions
- Supports VoIP implementation as per ED-137B of EUROCAE standard
- Ability to implement IP Spoofing for any network like Class C, Class B etc
- 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 different traffic options across simultaneous calls
- 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.
Click here for comprehensive information on supported codecs.
- RTP Voice Traffic Generation used in conjunction with SIP Signaling.
- Supports 64-bit RTP core to enhance performance - handles increased call rate of up to 3000 calls with high volume traffic.
- Supports both RTP G.711 Pass Through Fax Simulation (PKS200) and T.38 Fax Simulation over UDPTL (PKS211) simulation over IP.
- Transmit and receive pre-recorded video traces supporting video codecs like H.264, H.263, and VP8.
- Traffic impairments can be applied to messages to simulate error conditions that occur in real-time networks.
- Bulk Video call generation supported with H.264, H.263, and VP8 video codecs.
- User-defined statistics for RTP Voice and Video quality calls.
- User-defined voice quality statistics for received RTP Traffic can be calculated and updated periodically during run-time to a csv file
- Supports Secure Real-time Transport Protocol (or SRTP) traffic initialized over TLS (Transport Layer Security) or SSL (OpenSSL)
- Message Session Relay Protocol (MSRP) simulation supporting instant messaging. Supported call types include IM Only, Audio and IM calls, Video and IM calls.
Bulk Call Capability
- With normal RTP (PKS102 licensing) Maximum Simultaneous Calls - 2500, and Calls per Second - 250 (in high end server machines).
- Without RTP (only signaling) Maximum Simultaneous Calls - 70,000, and Calls per Second - 750 (in high end server machines).
- With MAPS™ HD RTP network appliance up to 20,000 endpoints per unit can be easily achieved (requires PKS109 and specialized hardware).
- Capability to generate more than 500 simultaneous video calls on a Core i7 systems.
- Automation, Remote access, and Schedulers to run tests 24/7
- Supported on Windows® 7, 8 or higher version operating systems.
- Supports 64-bit version to enhance signaling performance
- Detailed test result reports generation in PDF file format
- Option to send complete test report (traffic information and call events) to a central database, such as Oracle.
- Provides call quality metrics such as Listening MOS, Conversational MOS, PacketLoss, Discarded Packets, Out of Sequence Packets, Duplicate Packets, Delay and Jitter
- MSRP statistics including metrics such as Total MSRP Messages Sent, Received, Bytes Sent, and Bytes Received, which are calculated and updated periodically on run time
- MAPS™CLI interface based on a client-server model allows users to control all features of MAPS™ through APIs.
- Supported clients include TCL, Python, VBScript, Java, and .Net.
- Requires additional licenses
- 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 & loose routing, when requests are routed through proxies.
- Multi-protocol call trace for TDM / VoIP
- Testing NG9-1-1 emergency services and components within the ESInet.
Supported Protocols Standards
SIP Protocol Stack
ETSI TS 102-027-2 v4.1.1 (2006-07)
RFC 3262 - Reliability of Provisional Responses in the Session Initiation Protocol (SIP)
RFC 3311 - The Session Initiation Protocol (SIP) UPDATE Method
RFC 3455 - Private Header (P-Header) Extensions to the Session Initiation Protocol (SIP) for the 3rd-Generation Partnership Project (3GPP)
RFC 3515 - The Session Initiation Protocol (SIP) Refer Method
RFC 3310 - HTTP/SIP Digest Authentication Using Authentication and Key Agreement (AKA)
RFC 3263 - Session Initiation Protocol (SIP): Locating SIP Servers
Secure Real-time Transport Protocol (SRTP)
RFC 3711 - Secure Real-time Transport Protocol (SRTP)
RFC 3551 - Standard 65, RTP Profile for Audio and Video Conferences with Minimal Control
Message Session Relay Protocol (MSRP)
RFC 4975 - Message Session Relay Protocol (MSRP)
With MAPS SIP (PKS120), you can simulate complex call scenarios, modify SIP headers, automate test process, generate all types of traffic – Fax T.38, Voice, Video, Digits, Tones, and user-defined. Test scenarios can be manually executed using the Call Generation and Call Reception windows. The Load generation features allows for bulk call testing with parameters such as Call per second (CPS), Max Active Call, Minimum and Maximum Call Rates, Start Call Rates, and different statistical distribution patterns.
Following are typical applications where MAPS SIP tool is useful -
- Test PBX, Switch, Gateways/Routers
- Test IVR, Voice Mail, VoIP phones
- Generate/Accept tens & thousands of calls
- Voice, Fax, Digits, Tones, Data, Video traffic generation
- Automate Regression testing
- Feature testing
- Remote operation
- Scripting: Python, TCL, Java, VBScript Client APIs
- Voice, Data Quality, Quality of Service
Call Generation at UAC
Call Reception at UAS
UAC, UAS Conformance
MAPS™ SIP can be configured to act as UAC and sends request messages to conform UAS (DUT). MAPS™ SIP can also be configured at DUT to act as UAS, loaded with answer scripts to handle the incoming messages. UAS replies with 180 Ringing message for the initial INVITE request message from UAC, thus testing UAC.
Typical SIP Procedure
Call Generation at UAC
Call Reception at UAS
With the set of Proxy Conformance inbuilt scripts, MAPS™ can be configured to act as UAC and UAS simultaneously so that entire Proxy testing can be automated.
Here, MAPS™ acts as both UAS and UAC sending and receiving SIP messages while testing proxy (DUT). All the requests received from UAC (MAPS™) are replied back with the unmodified messages as seen in the message sequence window below.
DUT as Proxy Server
Proxy Conformance Testing
DUT as Registrant (Ex: PacketGen™) generates REGISTRATION SIP messages. The set of Registrant Conformance inbuilt scripts in MAPS™ tests to ensure that MAPS™ acts as Registrar and processes the received registration request messages from Registrant (DUT).
DUT as Registrant
Registrant Conformance Testing
Redirect Server Conformance
Receives requests and looks up the intended recipient of the request in the location database created by a registrar. MAPS™ acts as UAC and sends messages to test Redirect Server (DUT). Redirect Server is an entity that receives a request and sends back a reply containing a list of the current location of a particular user.
End-to-End- Gateway Testing
As shown in the figure below, MAPS™ is an ideal tool to evaluate Gateway / ATA product features such as call connectivity, call signaling, traffic generation, voice quality testing, codec, and hundreds of other features. For more details, contact GL Communications.
End-to-End Gateway Testing
Gateway Testing Call Scenario
Instant Messaging using SIP/MSRP
GL offers enhanced MAPS™ SIP emulator supporting Instant Messaging (IM) delivery conforming with RFC 4975/4976 - Message Session Relay Protocol (MSRP) protocol.
MAPS™ SIP software supports simulation of SIP/RTP sessions as well as instant messaging sessions over SIP using MSRP protocol (requires PKS112 licenses). The supported call types are IM Only Calls, Audio and IM Calls, and Video and IM Calls between multiple UAs.
MAPS™ SIP can simulate the end-points (SIP/RTP or SIP/MSRP User Agents) in an NG9-1-1 network and send and receive communications over ESInets. The software is capable of simulating high volume SIP sessions for load testing, and reporting thresholds.
MAPS™ SIP allows SIP vendors, wireless carriers, NG9-1-1service providers, and emergency communications centers to test IP applications for satisfactory working of NG9-1-1 services prior to deployment.
Unlike regular SMSs, IM requires a series of instant messages exchanged between two or more parties treated as a single conversational message session. Such a “session-mode” IM is possible using Message Session Relay Protocol (MSRP) where a series of related instant messages using SIP are treated similar to audio or video media sessions. MSRP supports multiple sessions to different end-points to be established over a single TCP connection.
Given below is a typical IM simulation between SIP/MSRP endpoints.
The following call generation screenshot depicts the MAPS™ SIP simulating ‘Audio and IM Calls’. During this call type, two different sessions will be established. There will be two media lines, one for audio using RTP session and another for instant messaging using MSRP session which are handled simultaneously.
Audio and IM Call Generation
MAPS™ SIP also provides user defined statistics feature, which can be configured to provide MSRP statistics including metrics such as Total MSRP Messages Sent, Received, Bytes Sent, and Bytes Received. These statistics are calculated and updated periodically on run time.
User Defined MSRP Statistics
Please Note: The XX in the Item No. refers to the hardware platform, listed at the bottom of the Buyer's Guide, which the software will be running on. Therefore, XX can either be ETA or EEA (Octal/Quad Boards), PTA or PEA (tProbe Units), UTA or UEA (USB Units), HUT or HUE (Universal Cards), and HDT or HDE (HD cards) depending upon the hardware.