SIP Trunking
Telephony industry standard - robust, enterprise-ready, and mandatory when you need call transfer, PBX integration, or managed platforms like LiveKit, VAPI, and Retell AI.
WebSocket Streaming
Developer-native path - highly cost-effective at scale, more direct, and ideal for custom AI pipelines built with Pipecat or bare-metal code.
Architectural Comparison
The two approaches operate at completely different layers of the stack. SIP is a telephony-layer protocol. WebSocket streaming is an application-layer transport.SIP Trunking Path
1
PSTN
Caller dials phone number
2
SIP Trunk (Vobiz)
PSTN → SIP INVITE routed to endpoint URI
3
Platform SIP Endpoint
LiveKit / VAPI / Retell terminates SIP, creates room
4
RTP Audio
UDP audio stream directly to platform
5
AI Agent
Receives WebRTC audio, runs STT → LLM → TTS
6
RTP Audio (back)
TTS audio back to caller via UDP
WebSocket Streaming Path
1
PSTN
Caller dials phone number
2
Vobiz Webhook
Fetches your webhook, receives VoiceXML stream directive
3
WebSocket (wss://)
Direct TCP connection established to your server
4
Your Server
Receives base64 µ-law audio directly from Vobiz
5
AI Pipeline
STT → LLM → TTS logic driven entirely by your code
6
WebSocket (back)
µ-law voice audio sent back over same socket connection
Important architectural nuance: These two architectures are not completely mutually exclusive. You can use a generic SIP Trunk provider to route a call to Vobiz, and then use a Vobiz VoiceXML
<Stream> directive to pipe that exact call to your custom WebSocket server. SIP handles the initial routing; WebSocket handles the audio layer.Full Decision Matrix
Platform Compatibility Matrix
Cost Analysis
The Vobiz channel rate is identical for both paths in spirit - the difference comes from whether you add a managed AI platform layer on top (SIP path) or own the pipeline yourself (WebSocket path). All pricing below is in INR.SIP Trunking Cost Stack
Vobiz base cost: ₹0.45/min + ₹500/month per number. Total = Vobiz rate + AI platform fees + STT/LLM/TTS API costs.
WebSocket Streaming Cost Stack
Vobiz WebSocket rate: ₹0.65/min + ₹500/month per number. Total = Vobiz rate + direct AI API costs only. No platform markup.
Latency Analysis
SIP has lower audio transport latency than WebSocket streaming. SIP uses UDP/RTP - a fire-and-forget protocol that never retransmits dropped packets, keeping audio delivery strictly real-time. WebSocket runs over TCP, which guarantees delivery by retransmitting lost packets - useful for data, but a source of jitter for live audio on poor networks. If latency is the only factor you care about, SIP wins. But latency is rarely why developers choose WebSocket streaming. They choose it for the ecosystem - direct access to AI frameworks (Pipecat), raw STT/LLM/TTS APIs, full pipeline control, and lower cost.When to Choose SIP
- You need to integrate with enterprise PBX (Avaya, Cisco UCM, Microsoft Teams Direct Routing)
- You’re using a managed AI platform (LiveKit, VAPI, Retell, ElevenLabs)
- You need maximum audio quality (G.722 wideband)
- Live call transfer must work without custom code
- You want carrier-layer scaling (add trunk channels without touching infra)
When to Choose WebSockets
- You’re building a custom AI pipeline (Pipecat, direct Python, Node.js)
- You want direct access to STT/LLM/TTS APIs
- Total cost matters more than platform abstraction
- You want full control of the pipeline (interruption, VAD, turn-taking)
- You’re comfortable scaling WebSocket workers yourself
Migration Path
You don’t have to commit to one architecture forever:- Start with SIP + managed platform for fastest time-to-market
- Validate product-market fit with low engineering investment
- Migrate to WebSocket streaming once volume justifies engineering ownership
- Use Vobiz
<Stream>directive to bridge: SIP trunk routes call → WebSocket pipes audio to your server