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)
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 other 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.
Users can remotely control MAPS™ using commands from the TCL environment. Multiple MAPS™ CLI servers can be controlled remotely from single client application (such as TCL, Python, VBScript, Java, and .Net). MAPS™ TCL Client application includes a MapsClientIfc interface, a packaged library that enables communication with the MAPS™ Server. TCL (Tool Command Language) Client is a command-line interface (TClsh85.exe) which is distributed along with MAPS™ Server application, using which any real-time scenarios can be simulated.
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.
- 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 (third-party added), attended call transfer, and call forwarding.
- 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 almost all industry standard codec types - G.711 (mu-Law and A-Law), G.722, G.729, G.726, GSM, AMR, EVRC, SMV, iLBC, SPEEX, and more. *AMR and EVRC 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.
- Test IVR, and Instant messaging features.
- 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.
- Supports Secure Real-time Transport Protocol (or SRTP) traffic initialized over TLS (Transport Layer Security) or SSL (OpenSSL)
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.
- The 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
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
RFC 3711 - Secure Real-time Transport Protocol (SRTP)
RFC 3551 - Standard 65, RTP Profile for Audio and Video Conferences with Minimal Control
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, CLI, TestShell, HP ALM
- 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
High Density RTP
Content Appended from maps hd..
Bulk Voice Traffic Simulation
RTP core can generate and receive voice traffic over IP networks and can work with applications such as GL’s MAPS™, VQuad™, and RTP ToolBox™ (PKS102). Transmit and Receive pre-recorded Voice Files in wave, pcm, and GL's proprietary pre-compressed GLW files with a synchronous Tx/Rx option. You can also directly send live voice using Talk using Microphone feature, and play the recorded voice files directly on to PC speakers. Some additional features that help in the voice traffic simulation are listed below-
- Allows to specify a desired voice payload type to each codec for sending and receiving payload;
- Sampling rate of the codec is displayed for the selected codec.
- Comfort noise generation is supported for A-law, µ-law and G.726 codecs for sending and receiving payload.
- Allows to set the buffer used for delayed packets that arrive at receiving end (both static and dynamic jitter buffers are supported)
- Allows to set QoS (Type of Service) properties such as precedence, delay, throughput and reliability values to the outgoing stream
- Comprehensive voice codec support -
- G.711 (A-law / Mu-law - 64kbps)
- G.711 App II (A-Law and Mu-Law with VAD Support)
- G.722.1 (32 k and 24 k)
- G.729, G.729B (8 kbps)
- G.726 (5 bit 40kbps/4 bit 32kbps/3 bit 24kbps/2 bit 16 kbps)
- G.726 (40/32/24/16 kbps with VAD)
- GSM 6.10 (13.2 kbps)
- GSM-HR (rate – 5.6kbps)
- GSM-EFR (rate - 12.2kbps, packet time fixed at 20msec.)
- SPEEX (Packet time fixed to 20msec)
- SPEEX_WB (Packet time fixed to 20msec)
- SMV (Modes - 0, 1, 2 and 3- Available if licenses are provided or owned, please call GL)
- AMR (4.75kbps, 5.15kbps, 5.9kbps, 6.7kbps, 7.4kbps, 7.95kbps, 10.2kbps, 12.2 kbps) (optional codec)
- EVRC (Rates - 1/8, ½ and 1) , EVRC0 (optional codec)
- EVRC_B (Rates - 1/8, ¼, ½, and 1), EVRCB0 (optional codec)
- EVRC_C (optional codec)
- H.263 video capture and conference capability
- H.264 video compression codec
For more details, please visit http://www.gl.com/voice-codecs.html
Bulk RTP FAX Simulation (T.30 pass through and T.38 UDPTL)
GL offers a variety of test tools to perform FAX over IP (FoIP) simulation and monitoring. Fax simulator supports both RTP G.711 Pass Through Fax Simulation (PKS200) and T.38 Fax Simulation over UDPTL (PKS211). Almost all MAPS™ IP based simulation products supports FAX simulation using any of these two methods. Typical applications of our Fax Simulators include load testing of fax servers, qualification testing of T.38 Gateways, testing of ATAs (Analog Terminal Adapters), testing of internet-aware fax machines, and many more.
With respect to channel capacity RTP pass-through supports up to 120 Fax ports, whereas T.38 fax simulation over UDPTL supports unlimited channels, and constrained only by CPU capacity.
Almost all MAPS™ IP products support fax simulation – MAPS™ SIP, MAPS™ SIP-I, MAPS™ MEGACO, MAPS™ BICC, MAPS™ GSM, and MAPS™ UMTS.
MAPS™ allows the user to initiate fax calls by sending call control messages using proper scripts and profiles. The profile allows necessary parameters of call control messages to be changed during runtime. The below figure depicts the T.30 fax call being generated using MAPS™ SIP.
GL’s RTP Fax Simulator simulates multiple fax calls over IP in T.30 pass through mode (using G.711 PCMU and PCMA). It can transmit pre-recorded Tiff image to DUT (Real-time Fax machine), receive Pass-Through fax from DUT, and record complete fax call messages as log file along with a Tiff image.
T.30 G.711 Pass through mode FAX simulation using MAPS™ SIP
MAPS™ SIP generates Re-Invite to switch from audio mode to image (FAX) mode. While the call in progress, the MAPS™ also provides events related to the progress of the call. After completion of the call, MAPS™ provide call quality statistics.
GL’s RTP Fax over UDPTL transport simulates multiple fax calls over IP using T38 protocol (compliant with ITU-T T.38 (03/2002)) up to maximum of 33.6 kbps speed.
The below figure depicts the T.38 fax call being generated using MAPS™ SIP.
T.38 Fax call in progress
Bulk Video Traffic Simulation
MAPS™ SIP provides the Bulk Video Call Simulation capability using its pre-recorded video traces supporting codecs like H.264, H.263, & VP8. On a high-performance computing platform (core-i7), it is possible to generate more than 500 simultaneous video calls. With a High Density (MAPS™ HD)platform, it is possible to achieve much more capacity. H.263 provides video capture and video conference monitoring capability, while H.264 is an industry standard codec for video compression. H.264 codec offers better compression performance over previous standards.
Below figure depicts the bulk video call simulation and RTP video transmission in MAPS™ SIP. Observe the decode part of the INVITE message showing the information about the audio and video codecs used in the request.
RTP/RTCP Packet Statistics
Statistics reports of RTP and RTCP packets transmitted on a session such as number of packets sent/received, dropped packets, out of sequence packets and more. Sender and receiver reports are also displayed using RTP/RTCP statistics applications.
RTP /RTCP Packet Statistics
Speech Quality Metrics (R Factor & MOS)
Quality metrics include various graphs for R-Factor and for MOS Factor. R Factor graph will display statistics such as, R-Listening, R-Conversational, R-G107 and R-Nom. MOS Factor graph will display statistics such as MOS CQ, MOS PQ and MOS Nom. The additional licensing (PKS108) RTP voice quality metrics for the received calls are calculated and are reported to MAPS™ application which are displayed including Listening and Conversational Quality MOS scores - MOS-LQ, MOS-CQ, and Listening and Conversational Quality R factors - R-LQ, R-CQ. Estimates are based on the ITU G.107 E Model.
A typical estimate of the MOS and R-Factor scores for each codec is available in www.gl.com/voice-codecs.html.
GL’s MAPS™ application provides a feature which helps users to customize the global statistics for RTP audio and video traffic. These global parameters are defined in the call generation scripts, which are calculated and updated periodically providing call quality metrics such as Listening MOS, Conversational MOS, PacketLoss, Discarded Packets, Out of Sequence Packets, Duplicate Packets, Delay and Jitter as shown in the figure below.
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.