Supported Protocols
The following protocols are supported:Supported Codecs
The following audio codecs are supported:G.711(preferred):A-law.U-law.
OPUS.G.722.
Features
Trunk management and Routing
- Transferring calls via SIP REFER or SIP INVITE.
- Multiple SIP trunks per customer.
- Configuring SIP trunks with options like tech prefix, SIP Diversion header, Outbound authentication (including
REGISTER). - Routing calls based on a trunk group, Direct Inward Dialing (DID), or DID range.
- Least-cost routing selection of outbound trunk.
Call Features
- Custom SIP headers on inbound and outbound calls.
- Mid-call SIP INFO requests.
- P-Asserted-Identity header to identify caller.
- Receiving compact SIP headers.
- Receiving re-INVITE with no Session Description Protocol (SDP).
- Answering machine detection.
- Configurable music on hold.
- Configurable Atmosphere Sounds playing in the background of the conversation.
- Configurable Silence Overlay playing background sounds when there’s a long pause in a call.
- TTS streaming to reduce latency.
Call Management
- Creating outbound calls via REST API.
- Performing operations on calls in progress via REST API.
STT and TTS providers
- Many cloud-based and on-premises speech providers are available, along with the option to integrate custom speech API tailored to your needs.
- Fallback speech configuration if the primary vendor fails.
Call Recording and Storage
- Call recording via integration with a third-party SIPREC server.
- Pausing or resuming transcription and recording to avoid the collection of Personally Identifiable Information (PII).
- Call recording with no third-party components, recordings streamed to the preferred cloud storage.
Call Quality Monitoring and Optimization
- Observability features include tracing call events with open telemetry.
- Visual view of speech latency (TTS and STT) on each call.
- Caching of TTS audio when possible to reduce costs.