MAPS™ SIP I Protocol Emulator
(Scripted SIP-I and SIP-T Protocol Emulation)
VoIP networks predominantly use SIP to setup and tear down voice calls and increasingly for video and multimedia calls. PSTN networks
predominantly use SS7 to do the same. PSTN SS7 signaling is quite different from SIP signaling and in many cases PSTN SS7 signaling may be
richer than SIP. There may be no one-to-one correspondence between SIP signaling messages and SS7 signaling messages. Also, it may not be
possible to enhance SIP to accommodate the additional features of SS7, and vice-a-versa.
Another consideration, the mobile network is evolving to an all IP network; hence it is advantageous to transport voice using SIP with SS7 routing
capabilities intact - thus the need for SIP to SS7 interworking. When a SIP-I is used to bridge the SS7 endpoints, the ISUP messages are carried
(encapsulated) along with SIP signaling messages.
GL’s Message Automation & Protocol Simulation™ (MAPS™) is an advanced protocol simulator/tester traffic generator for SIP-ISUP simulation over IP.
MAPS™ SIP I can simulate SIP-ISUP signaling specification as defined by the ITU / IETF standards ITU-T Q.1912.5.
MAPS™ product with SIP-I provides only the IP side of the Gateway at both ingress and/or egress depending on what one is trying to emulate.
In general, MAPS™ is a powerful Protocol Test platform, supporting a wide range of protocols including TDM, IP, and Wireless infrastructure protocols.
MAPS™ SIP I is designed for SIP-I Testing can simulate Signaling Gateway / Softswitch as a User Agents Client (UAC), which sends SIP requests
with ISUP message and User Agent Server (UAS), receiving and returning SIP responses with proper ISUP message attached.
The application is available as
- MAPS™ SIP I (Item # PKS126)
The MAPS™ SIP I supports powerful tools like Message Editor, Script Editor and Profile Editor which allow new scenarios to be created or existing
scenarios to be modified using SIP-I messages and parameters. It gives the flexibility of modifying any message so that we can easily duplicate the messages
generated by any entity to resolve interoperability issues.
With the purchase of RTP Core license (PKS102), MAPS™ SIP I 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.
GL’s MAPS™ SIP I is also available in High Density version (requires a special purpose network appliance and PKS109 RTP HD licenses). This is capable of high call intensity (hundreds of calls/sec) and high volume of sustained calls (tens of thousands of simultaneous calls/platform).
** Some of these traffic types requires additional licenses – contact GL for more information
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
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.
- Simulates Signaling Gateway, Softswitch as UAC, UAS, in the network.
- Supports transmission and detection of RTP traffic - digits, voice file, single /dual tones
- Handles Retransmissions and remote Retransmissions.
- Supports both UDP and TCP.
- Generates and processes SIP-I valid and invalid messages.
- Fully integrated, complete test environment for SIP-I.
- Supports complete customization of call flow and messages.
- Supports scripted call generation and automated call reception.
- Supports message templates for each SIP-I message and customization of the field values.
- Facilitates defining variables for the various protocol fields of the selected SIP-I message type.
- Supported on Windows® XP or higher version operating systems.
Supported Protocol Standards
||IETF RFC 3372
This window allows users to configure the necessary parameters to establish communication between MAPS™ SIP I and the DUT. The configuration
window for MAPS™ SIP I consists of SIP configuration for SIP ISUP interfaces elements. Default profile used to configure MAPS™ SIP-I with User Agent parameters.
Screen Shot of SIP-I Testbed Setup Configuration
Typical Call Flow Scenario
As shown below, the ingress gateway, the IP Network, and egress gateway provide a bridge for the SS7 signaling and traffic. SIP-I provides a
framework for the integration of ISUP with SIP. The ingress and egress gateways, at the point of interconnection provide the SS7 ISUP message
encapsulation and vice a versa. RTP is used to carry voice traffic as usual within the SIP network from gateway to gateway.
SIP-I Typical Call Flow
Call Generation option allows the user to simulate outgoing communications where an outgoing call is initiated by sending call control messages using
proper scripts and profiles. The profile allows necessary parameters of call control messages to be changed during runtime.
MAPS™ SIP I simulating UAC
As seen in the figure below, MAPS™ SIP I acts as UAC, initiating the call flow and receiving replies for the request messages from the DUT, thus
MAPS™ as UAS or UAC testing DUTs
For UAS testing, scripts are used to ensure MAPS™ SIP I act as UAC sending INVITE - INTIAL ADDRESS message. The response message is
received from the DUT (UAS) establishing media as seen in the message sequence window below.
Call Generation at UAC
Call generated from other entity can be automatically detected in call reception window by pre-setting the required scripts in the Incoming Call
MAPS™ SIP I Simulating UAS
In UAC testing, MAPS™ SIP I is loaded with a set of inbuilt scripts to handle the incoming messages. MAPS™ SIP I acts as UAS and sends
180 Ringing – ADDRESS COMPLETE response for the request message from DUT (UAC). Here, MAPS™ SIP I acts as UAS, and responds to messages
MAPS™ SIP I configured as UAS in SIP ISUP interface
Call Reception at MAPS™ SIP I (UAS)
Bulk Call Simulation
MAPS™ SIP I supports Bulk Call Simulation and Stress/Load Testing capabilities through Load Generation feature. Load Generation window helps
users configure Stress/Load Testing parameters such as Call per second (CPS) or Busy hour call attempts (BHCA), Max Simultaneous Calls and
Screen Shot of Load Generation Parameters
Customization of Call Flow and Messages Using Pre-Processing Tools
- Script Editor - The script editor allows the user to create / edit scripts and to define variables for the protocol fields. The script uses pre-defined
message templates to build call flow and perform send and receive actions. Script editor provides options to run the test for multiple iterations in sequential
or random flow. Commands allow retransmission of messages with specific interval. It also includes traffic commands to send and monitor voice, tones, digits, on the created sessions. It includes raw commands (send/monitor signaling bits, monitor power level, set idle code, and end task) to send WCS commands directly from MAPS™ to the server. .
Screen Shot of Script Editor
- Profile Editor - Profiles are used to provide the user configured values to the fields in the Messages (i.e., Message Template in
MAPS™ SIP I) through variables which are going to change during the course of a call. An XML file defines a set of multiple profiles with varying parameter values that allow users to configure call instances in call generation and to receive calls.
Screen Shot of Profile Editor
- Message Editor - The Message Template is a *.HDL file that comprises of protocol encoding parameters with preset values. It is required to
create a message template for every message in a protocol. The message templates are called within the scripts to perform scenario based testing.
Message Editor is used to create the ISUP part of the messages, while the SIP message templates are manually created.
Screen Shot of Message Editor for ISUP messages
Call Flow & Script Execution Control
Message Sequence - MAPS™ SIP I provides protocol trace with full message decoding, custom trace, and graphical ladder diagrams of
call flow with time stamp while simulation is running. Call flow graph allows easily verifying the messages exchanged between MAPS™ SIP I and DUT.
See Call Generation and Call Reception for details
Script Contents & Script Flow - The Scripts Contents window displays the contents of the script selected for call generation or reception. The
Script Flow window displays the set of statements successfully executed by MAPS™ SIP I to help users in troubleshooting a particular call scenario.
Events & Event Profile Editor - User-defined events allow redirection of script execution on-the-go. The custom parameters in the events can
also be changed during script execution using Event Profiles
Screen Shot of Script Contents and Script Flow
Call Statistics, Events, Link Status
Call Statistics & Status - By default, all call handling scripts (irrespective of the type of the functions) are assessed by MAPS™ SIP I to
provide statistical information about total calls, active calls, completed calls, passed calls, and failed calls. It is also possible to characterize the statistical
information under different groups of call handling scripts under a unique heading.
In addition, Call Generation and Call Reception windows provide useful call status & script execution results.
Screen Shot of Call Statistics and Status
Events Reporting – MAPS™ SIP I provide Event Log, Error Events, and Captured Errors windows that log the captured events and errors
encountered during the progress of the call.
Screen Shot of Events Log
Specifications are subject to change without notice.
Back to Message Automation & Protocol Simulation Index Page