i have been using elevenlabs for some time and i need to make fluent agents that work great in roma

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AI-generated Answer for m4agnum

Here’s how to configure a Romanian-first ElevenLabs agent that feels natural and low-latency.

1. Choose the right TTS model per goal

• Lowest latency: Flash v2.5 (supports Romanian). Use this when realtime feel matters most. (elevenlabs.io)
• Highest naturalness: Multilingual v2. Use this when quality/emotion is the priority. (help.elevenlabs.io)

2. Configure the agent for Romanian

• Add Romanian in Agent > Additional languages. When you add non‑English languages, the agent switches to the multilingual model family for those languages. (elevenlabs.io)
• Assign a Romanian voice in “language‑specific voices” for best pronunciation. Prefer voices trained for Romanian or clone a native Romanian voice. (elevenlabs.io)
• Enable the Language Detection system tool so the agent can switch automatically if callers mix languages. (elevenlabs.io)
• Optionally control speaking speed (e.g., 0.9–1.05) to match local cadence. (elevenlabs.io)

3. Make prosody sound human

• Use SSML break tags for natural pauses: <break time="1.0s" /> (up to 3s). These work across models (except Eleven V3). Don’t overuse. (help.elevenlabs.io)
• Note: SSML phoneme tags are English‑only; don’t rely on them for Romanian. (help.elevenlabs.io)

4. Cut perceived latency in production

• Use the WebSocket TTS API for bidirectional streaming; keep defaults for chunking and send flush: true at the end of each turn for timely audio start. (elevenlabs.io)
• Set auto_mode: true on the WS endpoint to reduce buffering delays. (elevenlabs.io)
• If your audience is in Europe or Asia, try the api-global-preview hostname for Turbo/Flash models to shave TTFB (preview; TTS only). (elevenlabs.io)
• Choose lean output formats for phone/web RTC: PCM 16 kHz or μ‑law 8 kHz (telephony‑optimized). (elevenlabs.io)
• Some voices incur more latency than others; default/synthetic/IVC are faster than Professional Clones. (elevenlabs.io)

5. API details that help with Romanian

• For short/ambiguous prompts, send language_code: "ro" (ISO 639‑1) so normalization/pronunciation rules are correct. Supported on WebSocket and REST TTS. (elevenlabs.io)
• Avoid mixing multiple languages in a single prompt; language is detected from text and mixed prompts can confuse selection. (help.elevenlabs.io)

6. If you transcribe users in real time

• Use Scribe v2 Realtime (auto language detection, VAD). Provide language_code "ro" if you already know the caller is Romanian. (elevenlabs.io)

7. Conversation flow settings that affect “MVP

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i am using a sip trunk

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AI-generated Answer for m4agnum

Great—SIP trunks are supported. Here’s how to wire yours to ElevenLabs Agents and get good real‑time performance.

SIP trunk essentials

• Inbound target: sip:sip.rtc.elevenlabs.io:5060;transport=tcp. Always include an identifier in the SIP URI, e.g. sip:[e-mail]:5060. Do not call [e-mail] without an identifier. (elevenlabs.io)
• Signaling transports: TCP or TLS (recommended). UDP is not available. Typical ports are 5060 (TCP) and 5061 (TLS). (elevenlabs.io)
• Codecs: use G.711 or G.722 (8 kHz / 16 kHz). Your SIP side must support one of these or resample. (elevenlabs.io)
• Media/RTP: allow UDP RTP (commonly 10000–60000). One‑way/no‑audio is usually firewall/NAT—verify RTP reachability and SRTP compatibility if you set media encryption to Required. (elevenlabs.io)
• Auth: Digest (username/password) is recommended; ACL/IP allowlisting is also supported. Enterprise can request static-IP SIP endpoints (sip-static.rtc.elevenlabs.io and regional variants) for allowlisting. (elevenlabs.io)
• Outbound from ElevenLabs: configure your trunk’s hostname/IP (no sip: prefix) plus transport and media‑encryption settings. (elevenlabs.io)
• Custom headers: you can pass X-CALL-ID and X-CALLER-ID; Twilio Elastic SIP Trunking is also supported via fallback header sip.twilio.callSid. (elevenlabs.io)
• Number formatting must match: if you dial with +E.164, import the number with +; if you dial without +, import without +. (elevenlabs.io)
• BYE/481 tip: if a TCP dialog reconnects, send BYE to the Contact URI from the 200 OK, not the load‑balancer hostname. (elevenlabs.io)

Romanian agent setup (fluency + latency)

• Language support: Romanian is supported across models; Multilingual v2 and Flash v2.5 include Romanian. (elevenlabs.io)
• Agent language config: adding additional languages switches the agent to the Multilingual v2.5 model by default. You can also configure language‑specific voices and first messages. (elevenlabs.io)
• Voice choice: pick a Romanian/RO‑accented voice (or clone/train with RO audio) for the most natural pronunciation; accent comes from the voice. (help.elevenlabs.io)
• Optional model override: with Multi‑voice support you can set model family per voice (Flash for lower latency vs. Multilingual for highest non‑English quality). Test which gives you the best RO flow. (elevenlabs.io)
• Turn‑taking: enable interruptions (barge‑in) and tune turn‑eagerness for faster conversational handoffs. (elevenlabs.io)

If you’re on Twilio

• You can use native Twilio number import (inbound+outbound) or Verified Caller IDs (outbound only). There’s

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i have tried low latency llm models and i saw when i moved to something quicker the transcribing quality got worse why is that so and how can i fix it

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AI-generated Answer for m4agnum

Short answer: switching to a “faster” LLM doesn’t improve (and can appear to worsen) transcription because ASR and the LLM are separate parts of the agent stack. What typically changes is turn‑taking behavior and how quickly you read “partial” vs “committed” transcripts.

Why quality can look worse

• In Agents, speech-to-text (Scribe), the LLM, TTS, and a turn‑taking model are separate components. Changing the LLM doesn’t change the ASR model itself. (elevenlabs.io)
• Scribe v2 Realtime streams predictive partial transcripts for low latency; these can be revised before they are “committed.” If your app consumes partials, it will look less accurate than committed segments. (elevenlabs.io)
• Faster LLM/“eager” turn‑taking can cause the agent to speak sooner, increasing overlaps and interruptions, which can degrade perceived transcript stability if you’re acting on partials. You can adjust turn eagerness to reduce early barge‑ins. (elevenlabs.io)
• Telephony specifics: SIP trunks often use G.711 at 8 kHz. If your STT session is configured for 16 kHz PCM while receiving 8 kHz μ‑law, resampling/mismatch can hurt accuracy. The Realtime STT supports ulaw_8000; the SIP trunk setup notes G.711/G.722 compatibility. (elevenlabs.io)

How to fix it (Romanian, SIP trunk)

• Use Scribe v2 Realtime for live calls and set model_id to scribe_v2_realtime on the Realtime Speech‑to‑Text WebSocket. Consume committed_transcript (or committed_transcript_with_timestamps) for downstream logic, not partial_transcript. (elevenlabs.io)
• Match the audio format to your trunk:
  • If you receive G.711 μ‑law at 8 kHz, set audio_format=ulaw_8000 and sample_rate=8000 in the STT session; or transcode server‑side to a supported PCM rate (e.g., 16 kHz) before sending. (elevenlabs.io)
  • Ensure your SIP integration is using a supported telephony codec (G.711 or G.722) per the SIP trunking guide. (elevenlabs.io)
• Lock the language to Romanian to avoid auto‑detect ambiguity: set language_code to “ron” (ISO‑639‑3) or “ro” (ISO‑639‑1). The API accepts either; Romanian is a supported language. (elevenlabs.io)
• Tune commit behavior instead of reading ultra‑early partials:
  • If you control commits, commit during natural pauses; committing every 20–30 seconds is a good practice. (elevenlabs.io)
  • If using VAD commit, avoid overly low silence thresholds that force premature segmenting; use the documented parameters (vad_silence_threshold_secs, vad_threshold, min_speech_duration_ms) and start from defaults. (elevenlabs.io)
• Provide previous_text context in your input_audio_chunk messages to stabilize predictions across segments or reconnects. (elevenlabs.io)
• Reduce early barge‑ins from the agent by setting turn eagerness to Normal/Patient in Conversation Flow settings so users can finish thoughts before the agent speaks. (elevenlabs.io)

If you share your current STT session params (audio_format, sample_rate, language_code, commit

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i ran out of credits i use the creator plan and i wanted to use the usage based payment method i dpnt understand now i have the credits that i set up there or is it on the next payment