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Flags

The following configuration flags are supported by the Cloudonix Mobile SDK for use with the setConfig() method:

  • TSX_TIMER - Default: 5000 Number of milliseconds to wait before re-transmitting a message that was not acknowledged.
  • ENABLE_ICE - Default: 1 Set to a positive number to enable ICE protocol negotiation for NAT traversal.
  • USER_AGENT - Default: Cloudonix/5 User agent string to send in SIP transactions.
  • LOG_LEVEL - Default: 6 Highest verbosity of log messages to propagate.
  • WEBRTC_LOG_LEVEL - Default: 0
  • PLAYBACK_LATENCY - Default: 140 Audio playback buffer length in milliseconds.
  • CAPTURE_LATENCY - Default: 100 Audio recording buffer length, in milliseconds.
  • ENABLE_TURN - Default: 0 Set to a positive number to enable TURN protocol negotiations for NAT traversal.
  • TURN_SERVER - Default: turn1.cloudonix.io The TURN server to use for NAT traversal.
  • TURN_PORT - Default: 3478 Port to use for the TURN protocol.
  • USE_TCP_FOR_TURN - Default: 0 Set to a positive number to force the use of TCP in TURN negotiations.
  • TURN_REALM - Default: asterisk Realm to use for TURN server authorization.
  • TURN_USER - No default Username to use for TURN server authorization.
  • TURN_PASSWORD - No default Password to use for TURN server authorization.
  • KEEP_ALIVE_INTERVAL - Default: 30 Number of seconds to wait between sending “keep alive” packet to keep NAT mappings open.
  • REGISTRATION_TIMEOUT - Default: 300 Number of seconds to set for a SIP registration expiry. The higher the number, the less network traffic is needed, but the more likely it is for a registration to get lost due to network change.
  • ENABLE_NAT - Default: 1 Set to a positive number to enable NAT detection.
  • USE_OPUS - Default: 1 Set to a positive number to allow the use of the Opus codec.
  • USE_ILBC - Default: 0 Set to a positive number to allow the use of the ILBC codec.
  • USE_G722 - Default: 0 Set to a positive number to allow the use of the G722 codec.
  • USE_G711 - Default: 0 Set to a positive number to allow the use of the PCMA and PCMU codecs.
  • MAX_RATE - Default: 24000 Maximum sample rate to be negotiated for audio.
  • MIN_RATE - Default: 8000 Minimum sample rate to be negotiated for audio.
  • DISABLE_REGISTRATION_REFRESH - Default: 1 Set to a positive number to prevent automatic re-registration when a registration expires.
  • ALLOW_MULTIPLE_CALLS - Default: 1 Set to a positive number to enable support for handling multiple calls at the same time.
  • NO_MEDIA_PERIOD - Default: 30 Number of seconds to allow a call to retry to reconnect the media.
  • ENABLE_STUN - Default: 0 Set to a positive number to enable the STUN protocol for detecting NAT.
  • STUN_SERVER - Default: no stun servers configured Set a STUN server address in the form of <ip>[:<port>] to use for STUN resolution. May be provided more than once to add multiple STUN servers to be tried.
  • DISABLE_SECURE_SIPS - Default: 1 Set to a positive number to disable using the “sips:” schema for signaling over TLS, which will force calls to drop if not all call legs are secure.
  • DISABLE_PLATFORM_LOGS - Default: 0 Set to a positive number to disable logging to the platform native log mechanism. With this flag set, logs will only be sent to the application logging event listener.
  • TLS_NEGOTIATION_TIMEOUT - Default: 1500 The timeout for establishing a TLS connection with the SIP server, in milliseconds. This timeout is used to detect network problems or firewalls blocking the TLS port, preventing the establishment of calls. This value cannot be set higher than the SIP_NEGOTIATION_TIMEOUT value.
  • SIP_NEGOTIATION_TIMEOUT - Default 3000 The timeout for establishing a SIP session with the SIP server - i.e. a SIP 100 provisional response was received - in milliseconds. This timeout is used to detect network problems or firewalls blocking a UDP port or another connection problem.
  • UNREGISTER_ACCOUNT_ON_NETWORK_ERROR - Default: false If during SIP account updates (after an account is already registered - if needed), the SIP stack encounter a network error - should the SIP client disconnect the account and send the onRegisterState(REGISTRATION_UNREGISTERED) event.
  • LOG_CALL_MEDIA_STATISTICS - Default: false During a call, the SDK will output RTCP summary statistics to the logs every 30 seconds.
  • MEDIA_RESET_TIMEOUT - Default: 4 seconds During a call, when the SDK detects that no media has been received for the specified number of seconds, it will attempt to reacquire the media by sending a re-invite to disable the media and then to enable it again. This should solve issues with overactive carrier firewalls or network outages in media relays (assuming the provider's media relay setup can recover by re-routing traffic to another service).