IP Protocol Test Suite

MAPS™ SIP can be used to simulate any interface of the VoIP network. A single MAPS™ 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 evaluation Gateway / ATA products for call connectivity, call signaling, traffic generation, voice quality testing, codec, and hundreds of other features.

MAPS™ SIP supports transmission and detection of various RTP traffic such as digits, voice file, Pass through FAX, single tone, and dual tones.

MAPS™ SIP Conformance Suite includes inbuilt basic and conformance scripts (*.gls) that allows itself to act as User Agent Client and to perform Proxy, Redirect Server, Registrar and UAS conformance testing.

MAPS™ ED-137 Radio, and Telephone emulators support VoIP implementation as per ED-137B of EUROCAE standard to simulate both Controller Working Group (CWP) and Radio Media Gateway (RMG) system.

MAPS™ ED-137 Recorder (PKS117) can simulate Recorder interface for both Air-to-Ground (AG) and Ground-to-Ground (GG) calls at Controller Working Position (CWP), Ground Radio Station (GRS) and Recorder endpoints as per ED-137/4B.

GL's PacketScan™ probes along VoIP Network Monitoring software is used to monitor packet flows in real-time from anywhere within a VoIP network. All major VoIP protocols are supported including SIP, H.323, Megaco, and MGCP.

Protocol analysis probes capture physical layer, signaling call flows, and voice/data traffic, and is sent to a central system comprised of a database engine, web server, and NetSurveyorWeb™, a web-based application, to facilitate data storage and retrieval through web browser clients.

Using GL's PacketScan™ thousands of calls can be monitored in real-time including detailed analysis of selected voice band streams. Applications include testing of IP phones, Gateways, IP Routers and Switches, and Proxies. Supports all most all industry standard protocols.

PacketScan™ is capable of monitoring /recording audio and video data of a session to files (in QuickTime *.qt format), allowing user to perform powerful data analysis. It also provides detailed call statistics such as packet loss, gap, jitter, delay, RTP performance statistics, R-factor & MOS scores, and unparalleled voice band statistics can be monitored simultaneously.

With MAPS™ RTP High Density (HD) network appliance (PKS109), MAPS™ SIP can achieve up to 20,000 endpoints per unit.) MAPS™ RTP High Density (HD) for IP/VoIP Platform supports SIP, SIP-I, MEGACO, MGCP, SIGTRAN, MAP, CAP, INAP, BICC, and similarly other protocols. This network appliance provides a modular and flexible solution to generate high volume voice calls using industry standard voice codes.

GL's MAPS™ or VQuad™ with Voice Quality Testing (VQT) software supports the next-generation voice quality testing standard for fixed, mobile and IP-based networks. Both MAPS™ SIP and VQuad™ allow to configure automatic call control (SIP protocol) with voice traffic. 

The VQT measurements are performed on this voice traffic as per ITU-T algorithms (POLQA, PESQ LQ/LQO/WB). The results are available in the standalone application as well as centralized web based application. Additional analytical results are displayed as part of the assessment such as MOS, E-Model, Signal Level, SNR, jitter, clipping, noise level, and delay (end to end as well as per speech utterance).

With RTP Speech Quality Metrics (PKS108), the received voice traffic is evaluated for quality analysis. The statistics are calculated based on the ITU G.107 E Model and reported to MAPS™ application. These include

• Listening and Conversational Quality MOS scores - MOS-LQ, MOS-CQ
• Listening and Conversational Quality R factors - R-LQ, R-CQ

PacketScan™ also performs detailed analysis of voice band streams gathering QOS statistics such as ITU G.107 E Model MOS, along with detail packet statistics, such as total packets, reordered, duplicate and missing packet counts, gap, jitter, and delay.

GL's MAPS™ SIP provides a means for Bulk Video Call generation, simulating many Video Calls from a single MAPS™ SIP system.  As part of this solution,  GL's PacketScan™ can provide real-time Video Call Analysis of the video generated over the SIP call using a non-intrusive connection to the network.

As a part of E2E test solutions, GL's VQuad™ Video Quality Testing solution comprises of end-to-end Video Conference testing, providing both Video and Voice MOS between two VAC Agents (one end could be an Android device).

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.

Analyzing Fax over IP (FoIP): GL's FaxScan™ and PacketScan™ can process V.34 SIP/T.38 and SIP/RTP PCAP captures in T.38 mode as well as transparent mode. Win PCAP captures can be processed from T.38 packets alone or as part of a capture file with multiple SIP calls.

GL's MAPS™ IMS test suite is capable of simulating multiple UEs and IMS core elements such as PCSCF, I-CSCF, S-CSCF, PCRF, MGCF. A complete IMS Network Lab environment along with LTE Test Suite can envisage for E2E testing of 4G LTE Network.

SIP-I signaling is used to bridge the SS7 endpoints, where the ISUP messages are encapsulated within SIP signaling messages. MAPS™ SIP I is designed for SIP-I Testing can simulate Signaling Gateway / Softswitch, which sends SIP requests with ISUP message and receive incoming SIP responses with proper ISUP message attached.

GL's MAPS™ (Message Automation and Protocol Simulation) platforms can be used to simulate various elements in a hybrid TDM VoIP network, and test interconnection of different networks, be it IP-IP, TDM-IP, or TDM-TDM. A typical depiction of a hybrid TDM/IP network, and its components simulated by MAPS™ is shown in the figure.