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06 Voice And Protocol Expansion ​

Scope:

  • Deepen the unified voice protocol entry
  • Land signaling/media providers
  • External recording/transcription providers
  • Reserve common voice protocol extensions

Constraints:

  • The voice business layer must not depend backward on concrete protocol DTOs
  • Signaling and media must stay layered

V1 protocol-end-to-end ​

  • [x] Add invite -> answer -> hangup end-to-end tests for the sip protocol entry
  • [x] Add invite -> dtmf -> hangup end-to-end tests for the sip-ws protocol entry
  • [x] Add invite -> transfer -> hangup tests for the pstn-provider webhook entry
  • [x] Add contract tests for /api/voice/protocols

Acceptance:

  • New protocol entries have full HTTP/runtime/voice chain verification, not just adapter unit tests

V2 call-control-semantics ​

  • [x] Add hold event mapping in the unified protocol entry
  • [x] Add resume event mapping in the unified protocol entry
  • [x] Add transfer event mapping to the voice use case in the unified protocol entry
  • [x] Clarify the dispatch strategy for DTMF across voice/conversation/automation

Current DTMF strategy:

  • Protocol adapters first normalize events into voiceprotocol.CallEventDTMF
  • The voice runtime receives and keeps the unified entry, and does not write call state directly
  • If conversation or automation later needs to react, it should subscribe to unified voice events instead of depending backward on concrete protocol DTOs

Acceptance:

  • Common call control events all flow through the unified normalization model

V3 provider-implementation ​

  • [x] Extract the recording provider from in-memory into a standalone mock/provider package
  • [x] Extract the transcription provider from in-memory into a standalone mock/provider package
  • [x] Add failure retry and error models for providers
  • [x] Reserve async webhook/callback write-back interfaces

Acceptance:

  • Recording and transcription providers can be swapped independently without affecting voice use cases

V4 media-topology ​

  • [x] Abstract RTP/SRTP adapter placeholders
  • [x] Clarify the WebRTC and RTP bridge contract
  • [x] Reserve conference/mixer capability interfaces
  • [x] Reserve QoS / voice quality metrics collection points

Acceptance:

  • Extending from point-to-point calls to more complex media topologies later will not require tearing down the current structure

V5 protocol-reservation ​

  • [x] Reserve voiceprotocol.ProtocolH323
  • [x] Reserve voiceprotocol.ProtocolFreeSWITCHEventSocket or similar PBX adaptation capability
  • [x] Reserve a hosted voice vendor webhook adapter contract
  • [x] Write a minimal implementation template document for onboarding new protocols

Acceptance:

  • The voice module is not implicitly designed as "SIP only"