Monitoring RTP audio streams by using RTCP

About audio stream monitoring

Voice Gateway uses the Real-time Transport Protocol (RTP) to send and receive audio streams from an end system, such as a SIP trunk. The RTP Control Protocol (RTCP) is part of the RTP specification (RFC 3550) and provides quality of service (QoS) statistics for RTP media streams. When RTCP is enabled, Voice Gateway uses the protocol to monitor RTP network data delivery metrics, such as jitter and packet transmission.

When audio stream monitoring with RTCP is enabled, Voice Gateway reports the following information:

• Network warnings: When a network-related incident occurs, such as high packet loss rate or a high average jitter in the RTP stream, the SIP Orchestrator and Media Relay log a warning. Voice Gateway also reports these warnings to the following places:

  • For each Watson Assistant turn, Voice Gateway reports warnings in the Watson Assistant turn event, if Watson Assistant turn event reporting is enabled. Voice Gateway also sends the warnings on the vgwNetworkWarnings API state variable, which is sent to Watson Assistant or service orchestration engine.
  • At the end of the call, the warnings are included in a warnings object in a call detail record event, if CDR event reporting is enabled.

• Network summaries: Voice Gateway provides a network summary in a rtpNetworkSummary object within the call detail record event. The network summary contains statistics from the inbound and outbound audio streams in the call, including average and maximum jitter and packet information.

Note that if the RTP end system does not support RTCP, outbound stream jitter and packet loss can't be reported.

Important: Enabling RTCP uses an additional UDP port per call. For each call, RTP uses a dynamic port within the range set on the RTP_UDP_PORT_RANGE environment variable, and RTCP uses the next port. For example, if RTP uses port 16384, RTCP will use port 16385. Make sure that these ports are open in your firewall. For more information, see Port settings for firewalls.

Enabling RTCP

To enable RTCP, set the following environment variables under the Media Relay container in your deployment configuration:

The following examples enable RTCP and change the default values for sending jitter and packet loss warnings.

• For Docker:

  media.relay:
    environment:
      ...
      - ENABLE_RTCP=true
      - RTCP_MAX_JITTER_THRESHOLD=45
      - RTCP_MAX_PACKET_LOSS_THRESHOLD=10
  

• For Kubernetes in IBM Cloud Kubernetes Service:

  "env": [
    ...
    {
      "name": "ENABLE_RTCP",
      "value": "true"
    }, {
      "name": "RTCP_MAX_JITTER_THRESHOLD",
      "value": "45"
    }, {
      "name": "RTCP_MAX_PACKET_LOSS_THRESHOLD",
      "value": "10"
    }]
  

Reported network information

When RTCP is enabled, Voice Gateway reports network warnings during the call and then aggregates the warnings and provides a network summary when the call ends.

"vgwNetworkWarnings" : [
  {
    "timestamp": "2018-02-08T13:10:01Z",
    "message": "CWSMR0035W: The inbound RTP audio stream packet loss of 7 percent exceeds the maximum packet loss threshold of 5 percent.",
    "id": "CWSMR0035W"
  }
]