SIP, Stream or Relay?

Compare SIP, Media Streams and Conversation Relay for connecting an AI voice agent to Indian phone numbers on Edesy - what you build, who runs the speech stack, and which to pick.

2 min read

Pick SIP if your agent already runs on a SIP-capable platform such as LiveKit Cloud, Vapi, Retell, Bolna or Dograh. Pick Media Streams if you run your own audio pipeline, for example in Pipecat, and want raw PCM over a WebSocket. Pick Conversation Relay if you only want to write the LLM logic and let Edesy do speech recognition, speech synthesis and barge-in.

Comparison#

SIP Media Streams Conversation Relay
What your side receives A SIP call (signalling + RTP audio) WebSocket, raw L16 PCM mono, 8 or 16 kHz WebSocket, JSON text
What you send back RTP audio Binary PCM frames + killAudio / disconnect text, play, sendDigits, language, end
Speech recognition Your platform You Edesy
Speech synthesis Your platform You Edesy
Voice activity detection and barge-in Your platform You Edesy (interruptible, min_bargein_word_count)
DTMF SIP / RTP dtmf event dtmf message (when enabled)
Inbound calls Yes, SIP target on the number Yes, media_stream target Yes, conversation_relay target
Outbound calls Your platform dials out through an Edesy SIP trunk stream_url on POST/v1/calls Not available today
Preflight test Test call to a saved SIP destination POST/v1/stream-test POST/v1/relay-test
Compatible with LiveKit Cloud, Vapi, Retell, Bolna, Dograh, xAI Grok voice agents, any SIP endpoint Pipecat (Edesy serializer), custom bots Bots written for Twilio ConversationRelay
Edesy charges Call minutes Call minutes Call minutes plus speech; ask for the rate
You pay separately Your agent platform Your STT, LLM and TTS vendors Your LLM
Docs SIP for AI voice agents Media Streams Conversation Relay

How to decide#

  1. Already on a voice-agent platform with a SIP option? Use SIP. Point the number's inbound target at your platform's SIP URI or a saved SIP destination, and for outbound create an Edesy SIP trunk for the platform to dial through. SIP TLS and SRTP are available on request. See SIP trunking.
  2. Running Pipecat or your own pipeline? Use Media Streams. You keep full control of the speech vendors and latency trade-offs.
  3. Want the least code? Use Conversation Relay. Your server only handles text, and an existing Twilio ConversationRelay bot ports with a URL change.

You can mix them: different numbers in the same workspace can use different targets, and you can change a number's target at any time with Inbound Routing.

Numbers for any of them#

All three work on Edesy's Indian virtual phone numbers: mobile, landline, toll-free and 92-series, bought self-serve after KYC. See Search and Buy.

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