This technology provides indication about the audio quality of the speech during enrolments and verifications in real-time via API and for future reference via logs. It is available for both single-server and multi-server deployments. Currently the Audio Quality Estimation feature is limited to polling (SIP, http streams) only, Webhooks support is planned to be added in the future. Based on…
Search: Sip
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…Elasticsearch docker.elastic.co/elasticsearch/elasticsearch oss:7.6.2 Apache License 2.0 Elasticsearch Curator bitnami/elasticsearch-curator:5 Apache 2.0 Filebeat docker.elastic.co/beats/filebeat-oss:7.6.2 Apache License 2.0 Grafana Apache license: Licensing | Grafana Labs Kibana docker.elastic.co/kibana/kibana-oss:7.6.2 Apache License 2.0 Logstash docker.elastic.co/logstash/logstash-oss:7.6.2 Apache License 2.0 OpenSSH server GNU license in repo: docker-openssh-server/LICENSE at master · linuxserver/docker-openssh-server Phonexia RTP proxy phonexia/voice-verify-rtp-proxy:1.12 Phonexia Phonexia SIP connector phonexia/sip-connector:v0.2.0 Phonexia Phonexia SPE phonexia/voice-verify-spe:3.31.0-dev-r3898-2 Phonexia Phonexia Voice…
…IP address (typically an IP address reservation on DHCP server) has to be reachable for API requests needs to be able to connect to a PBX (in case SIP calls are used) allowed ports TCP 22 – SSH connection TCP 80 – WebSockets, Kibana, Grafana TCP 5060 – SIP TCP 8000 – Voice Verify UDP 20000-20350 – RTP domain requirements…
…of net speech for enrollment; if the audio is longer, up to 60 seconds of net speech is used Voice Verify needs 3 – 5 seconds of net speech for verification; longer audio and speech length serve for detecting speaker change during whole call SIP calls Phonexia Voice Verify uses SIP protocol to register to a PBX and acts as…