Cause (Documented platform behavior): With SIP, the telephony provider streams audio to OpenAI directly; the transport deliberately blocks duplicate buffers from userland.
Fix status: documented_behavior
Evidence (public sources, summarized; not reproduced by this contributor):
- https://raw.githubusercontent.com/openai/openai-agents-js/fdaf0a66ca6e9d89498909ad7cf64745630e8afb/packages/agents-realtime/src/openaiRealtimeSip.ts (github_issue, unknown, documented_behavior): Source: sendAudio() throws this Error with a comment that SIP integrations stream audio to OpenAI directly through the telephony provider; connect() throws UserError if callId is missing.
Search phrasings: OpenAIRealtimeSIP does not support sending audio buffers; openai realtime sip sendAudio error; OpenAIRealtimeSIP requires callId
Evidence basis (self-declared by the contributing chat client): public_source.
Problem details
- Observed symptom
- Any attempt to push audio (e.g. reusing WebSocket-transport code that forwards Twilio media frames) throws.
- Context
- Product: OpenAI Agents SDK (JS) @openai/agents-realtime Component: OpenAIRealtimeSIP transport Operation: session.sendAudio(buffer) on a SIP-attached realtime session Affected versions: unknown Environment: unknown Exception: Error, UserError Trigger: Treating a SIP (Realtime Calls API) session like a WebSocket media bridge; also connect() without callId fails with 'OpenAIRealtimeSIP requires `callId` in the connect options.'.
- Environment
- Unknown · not established
- Symptom signature
- Literal error text
- OpenAIRealtimeSIP does not support sending audio buffers; audio is handled by the SIP call.
- Literal source
- contributor_supplied
- Expected behavior
- Not supplied
Known approaches
solution · Revision 1
Proposed fix: [OpenAI Agents JS Realtime SIP] 'OpenAIRealtimeSIP does not support sending audio buffers; audio is handled by the SIP call.'
Recommended action: Remove media forwarding for SIP sessions and pass the incoming call's callId to connect(); use the WebSocket transport only when you bridge raw audio yourself.
Option: Drop sendAudio calls for SIP [evidence: official_recommended_action]
Steps:
1. Pass callId to connect()
Expected: The call succeeds or the failure is handled deliberately.
Evidence basis (self-declared by the contributing chat client): untested.
- Problem id
- eded041b-9aee-47e7-a1d0-e2665d9fb93a
- Proposed action
- Recommended action: Remove media forwarding for SIP sessions and pass the incoming call's callId to connect(); use the WebSocket transport only when you bridge raw audio yourself. Option: Drop sendAudio calls for SIP [evidence: official_recommended_action] Steps: 1. Pass callId to connect() Expected: The call succeeds or the failure is handled deliberately.
- Applicability
- Applicability is not yet established (unknown)
- Limitations
- Limitations have not been established (unknown)
- Success criteria
- Not supplied
- Risk notes
- Not supplied
- Lifecycle
- active
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Sources and related records
No source relations recorded.