Max sip

INVITE and REGISTER tests in a single CallXML script (#2) Generating and receiving video+audio SIP calls Generate call: Receive call: Generate call with custom video RTP payload type: videoRtpmapEncodingName ="H264" videoPcapFileName ="sample_H264.pcap" videoSdpLines="a=fmtp:99 profile-level-id=42e00a; packetization-mode=1; max-br=452; max-mbps=11880" > Receive call with custom video RTP payload type: videoSdpLines="a=fmtp:99 profile-level-id=42e00a; packetization-mode=1; max-br=452; max-mbps=11880" /> Generating and receifing audio SIP calls with custom audio codec, with playback from .pcap file Generate call (in this example OPUS audio codec is used): Receive call (in this example OPUS audio codec is used): Using a custom formula to measure MOS score (different from G.107) Please use following CallXML scripts at call generator and receiver running at 2 different servers at 2 locations, to test quality of IP network between the servers. Script for call generator side: please change "192.168.10.50" to IP address of call receiver server. "10m" means max duration of call = 10 minutes. Script for call receiver: SIPREC: generate SIP calls with 2 RTP streams and play stereo WAV file The SIP Tester software simulates a SIPREC session recording client (SRC) and generates SIP calls to a session recording server (SRS), plays RTP streams from stereo WAV file. INVITE contains a mumtipart/mixed body with SDP and "recording-session" XML. The "recording-session" XML is configured in script. Content-Disposition: recording-session complete 2021-08-10T05:33:55 x3;encoding=hex true x x 2021-08-10T05:33:55 true sip:[email protected]:5060 x x;remote=00000000000000000000000000000000 y y 2021-08-10T05:33:55 false 218110433 separate 2021-08-10T05:33:55 218110434 separate 2021-08-10T05:33:56 ]]> Check if numbers from list are dialable (ringing) or not and save report to file

Opens a directory with log files. The files contain debugging information; by default error logs and GUI activity reports are sent to StarTrinity developers via encrypted channel. If you want to disable error reports, please set "DisableAnonymousReports" = "1" in settings tab. License information This screen displays information about license for current installation. For information about prices please contact sales manager by email. Command line interface (CLI) The SIP tester has a command line interface which is available with StarTrinity.SIPTester.CLI.exe. This executable runs in command prompt, it has following parameters: (if a parameter is not specified, it is taken from XML settings file) TotalCalls (required for active mode when generating calls) - total number of outgoing calls to generate, or 'infinite'. After exceeding this number, SIP Tester waits for current calls to complete TotalIncomingCalls (optional) - total number of calls to receive before exiting. After exceeding this number, SIP Tester waits for current calls to complete MaxCurrentCalls (optional) - max number of concurrent calls MaxCurrentCallsLimitMode (optional) - mode to limit max number of concurrent calls: 'incoming+outgoing' (default), 'outgoing', 'incoming' MaxTestDurationSec (optional) - max duration of test in seconds. If the "MaxTestDurationSec" is set, SIP Tester waits for all current calls to complete before exiting. Warning: please limit call duration by "maxansweredtime" to avoid hang calls IncomingCallsExitTimeoutSec (optional) - max time between incoming calls to stop test. Is used when you expect incoming calls to arrive during the test. If the "IncomingCallsExitTimeout" fires, SIP Tester waits for current calls to complete and exits with status -1 OutgoingCallsPeriodMs (optional) - interval between simulated calls in milliseconds NumberOfCallsPerBurst (optional, default is 1) - number of calls to generate per burst (tick) OutgoingCallsMode (optional) - mode of interval between simulated calls: fixed or random OutgoingCallXmlFile (optional) - CallXML file name to generate outgoing calls.

Software architecture Basic steps for active testing Basic steps for passive testing (monitoring) or VoIP recording Installation Main window UAC registrations UAS registrations Outgoing SIP calls Incoming SIP calls Current calls report: SIP information Current calls report: RTP information Call Detail Record (CDR) report Lowest quality calls Reports/Statistics Performance chart Stepwise testing Manual tests Impairments generation Settings Log License information Command line interface Web API Web API: UAC registrations Web API: Jobs Screen videos, training sessions Configuring SIP Tester with Cisco Unified Call Manager (CUCM) Report unclarityBasic steps for active testing Basic steps to run a simple active SIP stress test using graphical user interface (GUI) are: Install winpcap .NET Framework 4.5 and SIP Tester Configure UAC registrations Configure outgoing call stress parameters Configure CallXML scripts (test scenarios) via GUI or XML (optional) Configure audio files for playback and/or IVR audio verification, configure other settings Run a test (optional) If you make calls via internet, simultaneously run internet latency test, continuous speed test and VoIP readiness test to double-check internet connection Watch measured VoIP quality indicators in real time: Current calls - SIP indicators Current calls - RTP indicators Reports and statistics like "max jitter", "max packet loss" per call, etc. CDR report History charts (optional) configure email alerts and reports for call capacity overloads or call quality drops on settings screen (optional) listen to recorded audio files, or export results into .pcap files Alternatively you can use command line interface (CLI) with .bat scripts or windows service mode (StarTrinity.SIPTester.Service.install.bat) with Web API to run automated tests Basic steps for passive testing (monitoring) or VoIP recording In passive mode SIP Tester monitors all UDP packets on all network adapters like wireshark. It tries to interpret packets as SIP and RTP. There is no UDP port filter. To monitor SIP calls