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  • Overview
    • Introduction
    • Characteristics
    • Platforms
  • INITIAL Installation
    • 1. Instructions by platform
      • ProSBC Requirements Matrix
      • Baremetal Installation
        • List of Supported Network Interface Cards
        • SBC Certified Hardware
          • High Performance Baremetal Server
          • Medium Performance Baremetal Server
          • Ciena 3906mvi server for Customer Premises Equipment (CPE)
          • Qotom Barebone server for Customer Premises Equipment (CPE)
          • Lanner Barebone server for Customer Premises Equipment (CPE)
          • Telco Systems virtualization platform on Lanner NCA-2510 server for Customer Premises Equipment(CPE)
      • Virtual, self-hosted Installation
        • Virtual - Proxmox
        • Virtual - VMware
          • Launching an Instance of VMware vSphere
          • Deploying ProSBC on VMware
          • Adding Network Interfaces in VMware
          • Configuring Passtrough interfaces on VMware
      • Cloud Installation
        • Cloud - AWS
          • AWS Installation
          • Cloud Formation Installation
          • Instance Upgrade
          • AWS Additional Interface
          • AWS Installation Troubleshooting
          • Recovering an Elastic IP address
        • Cloud - Azure
    • 2. Initial Configuration
      • Initial Setup
        • SBC Management IP configuration
      • Basic configuration
        • Configuring IP interfaces
        • Creating a SIP stack
        • Creating a SIP transport server
        • Allocating an SIP NAP
        • Allocating a SIP open NAP
        • SIP Transport DNS settings
        • Creating a first call route
    • 3. Uploading a License
      • Host-control
        • SELinux
        • SELinux management
      • Add/Change Licenses
        • Add/Change Licenses Manually
  • Use Cases
    • Applications
      • Carrier Interconnection
      • Monitoring as a Service (MaaS)
      • NGN Interconnection
      • Operator Interconnection
      • SIP subscribe notify publish forwarding
      • STIR/SHAKEN
      • Transcoding
      • SIP Trunking
      • Hosted PBX
      • SIP Network Peering
      • Remote Workers
    • Interoperability Examples
      • STIR/SHAKEN with Transnexus and ClearIP
      • Fraud Detection [YouMail]
      • Skype Connect
      • Skype for Business S4B TCP
      • Skype for Business S4B TLS
      • Asterisk
      • 3CX
      • FreePBX
      • FusionPBX
      • FreeSWITCH
      • Twilio
      • Sippy
      • Avaya IP Office
      • Cisco UCM 12
      • Brekeke PBX
      • VitalPBX
      • Yeastar P-Series Cloud
      • VoIP.ms
      • Wildix
  • CONFIGURATION DETAILS
    • Configuration By Web Portal Category
      • System Settings
        • Setting the Role to Standalone
        • Setting the Role to a Primary Unit in a 1+1 System
        • Setting the Role to a Secondary Unit in a 1+1 System
        • Resetting the Host Role
        • Resetting the Network Device Role
        • Create Session Border Gateway Access Control List (ACL)
        • Session Border Gateway: Advanced Parameter Settngs
        • Create Session Border Gateway Access Control List (ACL) Filters
        • Connecting to the Web Server and Logging on to the Web Portal
        • Logging Off
        • Modifying Security Settings
        • Creating Web User groups
        • Creating Web Users
        • Modifying Web User Permissions
        • Enabling and Disabling a User
        • Deleting a User
        • Accessing Audit Logs
        • Activating the Configuration
        • Configuring a Web Portal Profile
        • Configuring the Date, Time, Timezone and NTP servers
        • Configuring the DNS
        • Create HTTP Service
        • Use HTTPS service
        • Configure HTTPS certificates
        • Configuring letsencrypt certificate
        • Configuring the ICMP
        • Configuring the SSH
        • Upgrade Telcobridges linux software packages
        • Retrieving a Software Release
        • Uploading a Software Release
        • Activating a Software Release
        • Retrieving a License
        • Uploading a License
        • Database Backup
        • Downloading a Database Backup
        • Uploading a Database Backup
        • Restoring a Database
        • Enabling the SNMP Agent
        • Configuring the SNMP Agent
        • Creating an SNMPv1/SNMPv2 Community
        • Creating an SNMPv3 User
        • Creating an SNMP Trap Destination
      • IP Network Settings
        • Configuring a Virtual Port
        • Configuring a VLAN
        • Configuring an IP Port Range
        • Configuring IP Interfaces
        • Configuring NAT Traversal
          • Local NAT Traversal
          • Remote NAT Traversal
        • DNS Configuration
          • Creating a DNS Local Entry
        • Configuring VoIP Interfaces
      • SIP
        • Creating a SIP Stack
        • Creating a SIP Transport Server
        • TLS/SRTP
          • Creating TLS Certificates
          • Adding TLS Certificates
          • Configuring TLS Profiles
          • List of Cipher Suites
        • Enabling SIP-I/SIP-T
        • SIPREC Forwarding
      • SIP Registrar
        • Creating a SIP Domain
        • Creating a SIP Registrar
        • Creating a SIP Register Filtering Rule
        • Creating a SIP Register Filtering Rule Condition
        • Creating a SIP Register Filtering Rule Action
      • Network Access Points (NAP)
        • Allocating a SIP Open Network Access Point (NAP)
        • SIP NAP Polling
      • NAP Profiles
        • Profile SDP Description
        • Fax Settings
          • Configuring Fax Relay
          • Configure Fax Passthrough
          • Configure Fax T38
          • Configure Fax NSE
          • Configure Fax VBD
      • Call Routing
        • Creating a First Call Route
        • Enable Flexible NOA Routing Script
        • Add NOA Columns in Routes
        • Import Customized Routing Script
        • Add Customer Column in Routes
        • Add Customized Filter Script To Main Script
        • Adding Label Routing to a Routing Script
        • Assign Routing Script Database Files to the Gateway Application
        • Add Digitmap Files to the System
        • Add Routeset Definition Files to the System
        • Assign Definition Digitmap Files on a per NAP Basis
        • Generate Dynamic Routes
        • Steps to configure label routing for Group of DIDs to a single outbound NAPs
        • Steps to configure label routing for Group of DIDs to multiple outbound NAPs
        • Group of DIDs to multiple outbound NAPs: Load-sharing mode
        • Group of DIDs to multiple outbound NAPs: Priority Mode
        • Update the Digitmap Files
        • Update the Routeset Definition Files
        • Configuring RADIUS Authorization
        • Importing a RADIUS Custom Dictionary
        • How to Use RegEx in Remapped Called and Calling Number Mask
          • Regular expression quick start guide
      • Lawful Intercept
        • Lawful Intercept Status
        • Verifying lawful interception
        • Importing a Lawful Interception .CSV File
        • Enabling Lawful Interception in a Routing Script
        • Configuring Lawful Interception
      • Call Detail Records (CDR)
        • CDR Variables
          • Call statistics format
        • Retrieve Text CDRs
          • Automatic CDR Retrieval
        • RADIUS CDRs
          • Configuring RADIUS
          • Adding RADIUS Server(s)
          • RADIUS CDR attributes
      • Routing Scripts
        • Development Guides & Tutorials
          • Accessing Routing Script Parameters
          • Parameter Mapping
          • Script Parameters Definition
          • Script Parameters Definition for SIP
          • Accessing Information about Registered Users
          • Route Parameters and Call Routing
          • Playing prompts, announcements, and tones
          • Recording
          • User-to-User Information
          • Radius Authorization
          • ENUM Query
          • DNS Query
          • Call Diversion Options
          • Call Transfer Requests
          • Redirection
          • Connect Number
          • Terminating Calls
          • NAP Status and other NAP Information
          • Telephony Services (CNAM Requests over SS7)
          • Custom User Context
          • Routing Script Tests
          • Create New Routing Script
          • Enable Routing Script
    • Configuration By Use Case
      • SIP Trunking Configuration
        • Configuration Files for SIP Trunking Scenario
        • SIP Trunk Configuration Instruction with 3CX
        • SIP Trunk Configuration Instruction with FreePBX
        • SIP Trunk Configuration Instruction with FusionPBX
        • SIP Trunk Configuration Instruction with FreeSWITCH
        • SIP Trunk Configuration Instruction with Twilio Elastic trunking
        • SIP Trunk Configuration Instruction with Avaya IP Office
        • SIP Trunk Configuration Instruction with Brekeke PBX
        • SIP Trunk Configuration Instruction with Avaya IP Office
        • SIP Trunk Configuration Instruction with Yeastar P-Series Cloud
        • SIP Trunk Configuration Instruction with Cisco UCM
        • SIP Trunk Configuration Instruction with VoIP.ms SIP trunking
        • SIP Trunk Configuration Instruction with Wildix Cloud VoIP PBX
        • Configuration for Adding ProSBC as a SIP Trunk in the FreePBX Server
        • FreePBX Extension Creation
        • FusionPBX SIP Trunk Creation
        • FusionPBX Extension Creation
        • FreeSWITCH SIP Trunk Creation
        • Twilio Elastic SIP Trunking Configuration
        • Sippy SIP Trunk Creation
        • Avaya IP Office Trunk Creation
        • Cisco UCM 12 Trunk Creation
        • Adding ProSBC as a SIP Trunk in the Brekeke PBX
        • VitalPBX Extension Creation
        • Adding ProSBC as a SIP Trunk in the Yeastar P-Series Cloud
        • Adding ProSBC as a SIP Trunk in the Wildix Cloud VoIP PBX
        • SIP Trunk Configuration Instruction with VitalPBX
        • VitalPBX SIP Trunk Creation
      • Configuring SIP Registration to SIP Proxy
      • Configuring a Hosted PBX
      • Multiple Domains/Hosted PBXs
      • SIP Network Peering / IP Carrier Interconnection
      • Remote Workers
        • Configuration Files for Remote Office/Workers
        • Remote Workers Configuration Instruction with FusionPBX
        • Remote Workers Configuration Instruction with 3CX
        • Remote Workers Configuration Instruction with FreePBX
        • Remote Workers Configuration Instruction with VitalPBX
      • ProSBC and ClearIP (TransNexus)
        • Configuration for STIR/SHAKEN with Transnexus' ClearIP service
        • Configuration for CNAM Verification and Robocall Analytics with Transnexus' ClearIP service
        • Configuration for Robocall Mitigation with Transnexus' ClearIP service
        • Configuration for 302 Redirect routing with Transnexus' ClearIP service
        • Configuration for CAPTCHA Authentication – 302 Redirect with Transnexus' ClearIP service
        • Configuration for STIR/SHAKEN with Transnexus' ClearIP service
      • Transcoding Unit Configuration
        • Baremetal and Virtual Machine
        • Show the Hardware Units menu
        • Adding Transcoding Unit
      • Configuration for Adding YouMail Script to Routing Scripts
      • Skype Connect Example Configuration
      • Skype for Business Example Configuration
      • 3CX Phone Provisioning Configuration
        • Configuration for 3CX PBX Server with the ProSBC to receive T38 Faxes
        • Configuration for 3CX PBX Server with the ProSBC as SIP trunk
      • SIP Emergency
      • SIP registration forwarding
        • Creating a SIP Domain
        • Configuring SIP Registration for Open NAP
        • Configuring SIP Registration for regular NAP
      • RTP no-anchoring
        • Parameter: Allow low-delay media relay
          • Configuring an IP Port Range
        • Creating Profiles
          • Modifying SDP Profile Settings
          • Modifying SIP Profile Settings
          • Modifying RTP and Audio Settings
          • Modifying FAX Relay Profile Settings
          • Modifying Telephony Profile Settings
          • Modifying Tones and Call Progress Options
          • Modifying IVR Record Profile Settings
          • Modifying LNP Profile Settings
          • Modifying Multilevel Precedence and Preemption (MLPP) Options
          • Modifying Call Transfer Profile Settings
          • Modifying Tone Definition Profile Settings
    • Configuration Parameters (all)
    • Routing Script - SIP 302 Handling
  • Maintenance & Troubleshooting
    • Maintenance Guide
      • Check Disk Space
      • ProSBC Processor Usage
      • Troubleshooting Toolpack
      • Restoring a Database
    • System Upgrades
      • Migrate current database
      • Upgrade Telcobridges linux software packages
    • Software version release notes
    • Software version release download
    • ProSBC public roadmap
  • Troubleshooting & Support
    • Troubleshooting Tips & Actions
      • Configuring Call Trace
        • Retrieving Call Trace
        • Call Trace Filter Parameters
      • Creating a test call
      • tbsigtrace: Signaling trace capture tool
        • Accessing Device
          • TMG:Change Management IP Address
          • Password less ssh
          • How to setup ssh tunnel with PuTTY
        • Live Signaling Capture with tbsigtrace
      • How to Get Rid of Sub Optimal Warning
      • How to Lower The Trace Level on an Application
      • TBReport
      • VoIP Ethernet Capture on a ProSBC
      • Enabling Call Recording
      • Accessing the Call Recording
      • Routing Scripts
        • Update Your Routing Scripts
        • Disabling a Call Route
    • Troubleshooting Common Problems
    • Support Links
      • Support Forums
      • ProSBC Training
      • Customer Dashboard User Guide
      • Contacting TelcoBridges technical support
      • Frequently Asked Questions
      • Sending Large Files to TelcoBridges
    • How to use tbx cli tools remote program
  • Tools, Tips, and Tricks
    • TelcoBridges Magic Bookmark
    • Video Library
    • RESTful API
      • Postman Examples
      • Ruby Examples
      • TBConfig Examples
        • Exporting a Configuration
        • Importing a Configuration
        • Activating a Configuration
        • Updating a Route
        • Dropping Calls
      • ProSBC:Restful API SIP Domain
      • ProSBC:Restful API SIP Domain Registrar
      • Extracting Call Traces with the API
    • TBStatus API
      • Tbstatus monitoring
      • Status API
      • Dropping calls
  • Appendices
    • Appendix A: Glossary
      • Glossary: Call Detail Records (CDR)
      • Glossary: Call routing
      • Glossary: DNS
      • Glossary: Mean Opinion Score (MOS)
      • Glossary: NAP
      • Glossary: RADIUS
      • Glossary: Ringback tones
      • Glossary: SAP
      • Glossary: Signaling protocols
      • Glossary: SIP
        • Glossary: Route retry
        • Glossary: SIGTRAN
        • Glossary: SIP-I/SIP-T
        • Glossary: SIP gateway
        • Glossary: SIP Registration
      • Glossary: Softswitch
      • Glossary: Toolpack
        • Glossary: Web server
        • Glossary: tboamapp
          • Glossary: Tbtoolpack Service
            • System Id
              • Gateway Port
          • Primary/Secondary
          • Master/Slave
            • Active/Standby
      • Glossary: Unified communications
      • Glossary: Web Portal
      • Glossary: DTMF Relay
    • Appendix B: Product Datasheets
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  1. CONFIGURATION DETAILS
  2. Configuration By Web Portal Category
  3. Call Detail Records (CDR)
  4. CDR Variables

Call statistics format

Field name
Description

Statistics

@{StatSummary}

Summary of the RTP and T.38 stats in a pre-defined format: RTP( Rx(Pkt=2 CN=0 Signaling=0 Err=0) Tx(Pkt=0 CN=0 Signaling=0 Err=0)) T38( Rx(Pages=0 Pkt=0 Err=0) Tx(Pages=0 Pkt=0)) MOS(In=4.3 Out=4.3}) NetworkQuality(In=100 Out=100) For customized format, use other variables below.

MOS statistics

@{Stat:MOS}

Summary of the MOS stats: (Codec;Ingress;Egress)

@{Stat:MOS:Codec}

Maximal MOS value for the audio codec used

@{Stat:MOS:Ingress}

Calculated MOS value using RTP RX statistics

@{Stat:MOS:Egress}

Calculated MOS value using RTCP statistics (available only when RTCP is enable)

@{Stat:MOS:Global}

Minimum value between MOS ingress and MOS egress

Network quality statistics

@{Stat:NetworkQuality}

Summary of the network quality stats: (Ingress;Egress)

@{Stat:NetworkQuality:Ingress}

Calculated network quality percentage from 0 (bad) to 100 (perfect) for packets received

@{Stat:NetworkQuality:Egress}

Calculated network quality percentage from 0 (bad) to 100 (perfect) for packets transmitted (available only when RTCP is enable)

@{Stat:NetworkQuality:Global}

Minimum value between ingress and egress

Receive statistics

@{Stat:Rtp:Rx}

Summary of the Rx stats: (Packets;CNPackets;SignalingPackets;Errors;MOS:Ingress)

@{Stat:Rtp:Rx:Packets}

Number of packets received from the IP network (audio + comfort noise + signaling)

@{Stat:Rtp:Rx:VoicePackets}

Number of voice (audio) packets received from the IP network

@{Stat:Rtp:Rx:VoiceBytes}

Number of payload bytes from voice (audio) packets received from the IP network

@{Stat:Rtp:Rx:CNPackets}

Number of comfort noise packets received from the IP network

@{Stat:Rtp:Rx:SignalingPackets}

Number of signaling packets (like "Telephony events") received from the IP network

@{Stat:Rtp:Rx:IPPackets}

Number of packets received from the IP network

@{Stat:Rtp:Rx:IPBytes}

Number of bytes received from the IP network

@{Stat:Rtp:Rx:RFC2833Tones}

Number of RFC2833 tones received from the IP network, and played back to TDM

@{Stat:Rtp:Rx:RedundantPackets}

Number of redundant packets received from the IP network (same sequence number received more than once)

@{Stat:Rtp:Rx:MeanPower}

Mean audio power (from IP to TDM direction), in dBov (0 the loudest, negative is softer level). Available only for calls with Echo cancellation enabled, thus only TDM<->IP calls.

@{Stat:Rtp:Rx:VoiceDuration}

Duration (in seconds) of voice (audio), from IP to TDM direction

@{Stat:Rtp:Rx:PlayoutDelay}

Current IP to TDM audio delay in millisecond (adaptive jitter buffer delay)

@{Stat:Rtp:Rx:MaxPlayoutDelay}

Maximum (for the whole call) IP to TDM audio delay in millisecond (adaptive jitter buffer delay)

@{Stat:Rtp:Rx:SsrcChanges}

Number of changes of "SSRC" RTP header value (counting initial SSRC received as one)

@{Stat:Rtp:Rx:Concealments}

Number of times decoder made audio concealment to preserve audio continuity (frequent upon first packet of a call, or after RFC2833 tones)

@{Stat:Rtp:Rx:Echo:ERLLevel}

Attenuation of a signal from the received-out port to the send-in port of an echo canceller

@{Stat:Rtp:Rx:Error:BufOverflows}

Number of packets dropped due to jitter buffer adaptation (to keep play-out delay low)

@{Stat:Rtp:Rx:Error:SeqErrors}

Number of sequence errors detected among packets received from the IP network

@{Stat:Rtp:Rx:Error:BadPackets}

Number of packets received with invalid header, or other error preventing proper audio decoding

@{Stat:Rtp:Rx:Error:RFC2833BadPackets:

Number of RFC2833 signaling packets that could not be properly decoded

@{Stat:Rtp:Rx:Error:LatePackets}

Number of packets that arrived after the moment they should have started to be played-out to TDM

@{Stat:Rtp:Rx:Error:LostPackets}

Number of packets considered "lost" (never received for play-out to TDM)

@{Stat:Rtp:Rx:Errors}

Total of all "Rtp:Rx:Error:" statistics above

Transmit statistics

@{Stat:Rtp:Tx}

Summary of the Tx stats: (Packets;CNPackets;SignalingPackets;Errors;MOS:Egress)

@{Stat:Rtp:Tx:Packets}

Number of packets sent to the IP network (audio + comfort noise + signaling)

@{Stat:Rtp:Tx:VoicePackets}

Number of voice (audio) packets sent to the IP network

@{Stat:Rtp:Tx:CNPackets}

Number of comfort noise packets sent to the IP network

@{Stat:Rtp:Tx:SignalingPackets}

Number of signaling packets (like "Telephony events") sent to the IP network

@{Stat:Rtp:Tx:IPPackets}

Number of packets sent to the IP network

@{Stat:Rtp:Tx:IPBytes}

Number of bytes sent to the IP network

@{Stat:Rtp:Tx:RFC2833Tones}

Number of RFC2833 tones sent to the IP network (relaying corresponding detected TDM tones)

@{Stat:Rtp:Tx:MeanPower}

Mean audio power (from TDM to IP direction), in dBov (0 the loudest, negative is softer level). Available only for calls with Echo cancellation enabled, thus only TDM<->IP calls.

@{Stat:Rtp:Tx:VoiceDuration}

Duration (in seconds) of voice (audio), from TDM to IP direction

@{Stat:Rtp:Tx:Echo:ACOMLevel}

Attenuation of a signal from the receive-out port to the send-out port of an echo canceller

@{Stat:Rtp:Tx:Error:LostPackets}

Number of packets considered "lost" by remote side (only available if RTCP is used)

@{Stat:Rtp:Tx:Error:ArpFailure}

Number of failures to ARP remote IP address to send RTP packets to (preventing sending any packets)

@{Stat:Rtp:Tx:Errors}

Total of all "Rtp:Tx:Error:" statistics above

RTCP statistics

(only if RTCP is enabled on the profile)

@{Stat:Rtcp}

Summary of the Rtcp stats: (Packets;LostPackets;Jitter)

@{Stat:Rtcp:Packets}

Number of packets reported as received by remote equipment

@{Stat:Rtcp:LostPackets}

Number of packets reported as lost by remote equipment

@{Stat:Rtcp:Jitter}

Inter-packet interval jitter, in millisecond

@{Stat:Rtcp:RoundTripDelay}

Round-trip delay in milliseconds. Requires extended RTCP (XRTCP) to be enabled.

T38 statistics

@{Stat:T38:StateChanges}

Number of T.38 state changes, through T.30 negotiation between FAX machines

@{Stat:T38:Rx

Summary of the T.38 Rx stats: (Pages;Packets;Errors)

@{Stat:T38:Rx:Pages}

Number of FAX pages received from T.38 packets from the IP network

@{Stat:T38:Rx:Packets}

Number of T.38 packets received from the IP network

@{Stat:T38:Rx:Bytes}

Number of T.38 payload bytes from T.38 packets received from the IP network

@{Stat:T38:Rx:Duration}

Duration of the T.38 session from IP to TDM

@{Stat:T38:Rx:PlayoutDelay}

Current IP to TDM audio delay in millisecond (re-modulation and jitter buffer delay)

@{Stat:T38:Rx:MaxPlayoutDelay}

Maximum IP to TDM audio delay in millisecond (re-modulation and jitter buffer delay)

@{Stat:T38:Rx:Error:BufOverflows}

Number of packets dropped because "from IP" jitter buffer is full

@{Stat:T38:Rx:Error:BufUnderflows}

Number of times T.38 re-modulation to TDM had no data (received from IP) to re-modulate

@{Stat:T38:Rx:Error:SeqErrors}

Number of sequence errors detected among packets received from the IP network

@{Stat:T38:Rx:Error:BadPackets}

Number of packets received with invalid header, or other error preventing proper audio decoding

@{Stat:T38:Rx:Errors}

Total of all "T38:Rx:Error:" statistics above

@{Stat:T38:Tx}

Summary of the T.38 Tx stats: (Pages;Packets)

@{Stat:T38:Tx:Pages}

Number of FAX pages received from TDM, sent to T.38 packets to the IP network

@{Stat:T38:Tx:Packets}

Number of T.38 packets sent to the IP network

@{Stat:T38:Tx:Bytes}

Number of T.38 payload bytes from T.38 packets sent to the IP network

@{Stat:T38:Tx:Duration}

Duration of the T.38 session from TDM to IP

@{Stat:T38:FromTdm:MeanPower}

Mean audio power received from TDM side, in dBov (0 the loudest, negative is softer level)

TMS-IP statistics

(only for calls joined to a call leg on another adapter, and that used TMS-IP to carry audio from one adapter to the other)

@{Stat:TmsIp}

Summary of the TMS-IP stats: (Packets;Errors)

@{Stat:TmsIp:Packets

Number of TMS-IP packets received from the other joined call leg

@{Stat:TmsIp:Bytes

Number of TMS-IP bytes received from the other joined call leg

@{Stat:TmsIp:Duration

Duration (in seconds) of the TMS-IP connection

@{Stat:TmsIp:Error:BufOverflows

Number of TMS-IP packets dropped due to jitter buffer overflow

@{Stat:TmsIp:Error:SeqErrors

Number of TMS-IP packet sequence errors detected among packets received from the IP network

@{Stat:TmsIp:Error:BadPackets

Number of TMS-IP packets received with invalid header, or other error preventing proper audio decoding

@{Stat:TmsIp:Error:LatePackets

Number of TMS-IP packets that arrived too late, causing jitter buffer underrun

@{Stat:TmsIp:Error:LostPackets

Number of TMS-IP packets considered "lost" (never received)

@{Stat:TmsIp:Errors}

Total of all "TmsIp:Error:" statistics above

Field name
Description

Receive statistics

@{ErrorSummary}

Summary of reported leg errors in a pre-defined format: FromNetwork(NoPackets;ExcessivePackets;ExcessivePayloadTypeChanges;ExcessiveIpPortChanges;ExcessiveSsrcChanges) ToNetwork(NoPackets) For customized format, use other variables below.

@{Error:FromNetwork}

Summary of the leg errors 'from network': (NoPackets;ExcessivePackets;ExcessivePayloadTypeChanges;ExcessiveIpPortChanges;ExcessiveSsrcChanges)

@{FromNetwork:NoPackets}

When '1', no packets received from the network in the last few seconds, while packets were expected.

@{Error:FromNetwork:ExcessivePayloadTypeChanges}

When '1', excessive payload type changes were detected in the last few seconds. Payload type changes will be ignored. Packets may be dropped.

@{Error:FromNetwork:ExcessiveIpPortChanges}

When '1', excessive remote IP/Port changes were detected in the last few seconds. Received packets will be dropped.

@{Error:FromNetwork:ExcessiveSsrcChanges}

When '1', excessive remote SSRC changes (SSRC field in RTP packets header) were detected in the last few seconds. Received packets will be dropped.

@{Error:FromNetwork:ExcessivePackets}

When '1', excessive packets/bytes rate was received from the network. All packets will be dropped.

@{Error:ToNetwork}

Summary of the leg errors 'tonetwork': (NoPackets)

@{Error:ToNetwork:NoPackets}

When '1', no packets were sent to network in the last few seconds, while we were supposed to send packets (remote IP not reachable?)

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Last updated 8 months ago

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