> ## Documentation Index
> Fetch the complete documentation index at: https://docs.convocore.ai/llms.txt
> Use this file to discover all available pages before exploring further.

# Understanding the SIP Server

> Learn how Convocore's SIP server bridges your calls to AI agents

# Understanding the SIP Server

Convocore provides a fully managed SIP server that bridges calls from your SIP trunk provider to your AI agents. This page explains how it works behind the scenes.

## What is the SIP Server?

The SIP server (running Asterisk) is the communication bridge that:

* **Receives calls** from your SIP trunk provider
* **Processes audio** in real-time
* **Routes calls** to the correct AI agent
* **Manages the connection** between telephony and AI

**\[IMAGE PLACEHOLDER: Diagram showing SIP server's role in the call flow]**
*Filename suggestion: `asterisk-role-diagram.png`*
*Description: Architecture diagram showing: SIP Provider → Convocore SIP Server (with boxes for SIP Handler, Audio Processor, Call Router) → Convocore AI Agent*

<Note>
  **Important**: The SIP server is fully managed by Convocore. You don't need to
  install, configure, or maintain anything.
</Note>

## Server Details

Your SIP provider should route calls to:

```
IP Address: 134.209.121.168
Port: 5060
Protocol: UDP (SIP)
```

## How It Works

### Call Flow

When someone calls your phone number:

```mermaid theme={null}
sequenceDiagram
    participant Caller
    participant SIP Provider
    participant Convocore SIP Server
    participant AI Agent

    Caller->>SIP Provider: Dials Phone Number
    SIP Provider->>Convocore SIP Server: Routes Call via SIP
    Convocore SIP Server->>AI Agent: Establishes Connection
    Note over Convocore SIP Server,AI Agent: Bidirectional Audio Stream
    AI Agent->>Caller: Responds in Real-Time
    Note over Caller,AI Agent: Conversation Continues
    Caller->>Convocore SIP Server: Hangs Up
    Convocore SIP Server->>AI Agent: Terminates Connection
```

### Step-by-Step Process

<Steps>
  <Step title="Call Arrives">
    Your SIP provider receives an incoming call and routes it to
    `134.209.121.168:5060`
  </Step>

  <Step title="Number Lookup">
    The SIP server looks up the phone number in Convocore's database to find the
    assigned agent
  </Step>

  <Step title="Agent Connection">
    The server establishes a WebSocket connection to the assigned AI agent
  </Step>

  <Step title="Audio Bridge">
    Audio flows bidirectionally: - Caller's voice → SIP Server → AI Agent - AI
    Agent's response → SIP Server → Caller
  </Step>

  <Step title="Real-Time Processing">
    The AI agent processes speech, generates responses, and speaks back through
    the bridge
  </Step>

  <Step title="Call Termination">
    When either party hangs up, the connection is cleanly terminated
  </Step>
</Steps>

## Technical Architecture

### Components

<CardGroup cols={2}>
  <Card title="SIP Handler" icon="phone">
    Manages SIP signaling (call setup, teardown, etc.)
  </Card>

  <Card title="Audio Engine" icon="waveform">
    Processes PCM16 8kHz mono audio streams
  </Card>

  <Card title="Call Router" icon="route">
    Routes calls to the correct AI agent based on phone number
  </Card>

  <Card title="WebSocket Bridge" icon="plug">
    Connects to Convocore's AI agent infrastructure
  </Card>
</CardGroup>

### Audio Processing

The server handles audio format conversion:

* **Input**: Various codecs from your SIP provider (PCMU, PCMA, etc.)
* **Processing**: Converts to PCM16 8kHz mono
* **Output**: Sends to AI agent in optimal format
* **Response**: Converts AI agent audio back to SIP-compatible format

### Supported Codecs

The SIP server supports standard telephony codecs:

* ✅ **PCMU (G.711 μ-law)** - North America standard
* ✅ **PCMA (G.711 A-law)** - International standard
* ✅ **Opus** - High-quality, low-latency
* ✅ **G.722** - Wideband audio

## Network Requirements

### For Your SIP Provider

Your SIP provider needs to:

<Checklist>
  * [ ] Support outbound SIP trunk routing - \[ ] Allow routing to custom IP
    addresses - \[ ] Support UDP on port 5060 - \[ ] Have compatible codec support
    (PCMU/PCMA minimum) - \[ ] Not require IP whitelisting (or whitelist
    134.209.121.168)
</Checklist>

### Firewall Rules

The Convocore SIP server has proper firewall rules configured:

* **Inbound SIP:** Port 5060 (UDP) - Open to SIP providers
* **Inbound RTP:** Ports 10000-20000 (UDP) - For audio
* **Outbound:** HTTPS/WSS to Convocore AI infrastructure

You don't need to configure any firewall rules on your end - just point your SIP provider to the server.

## Server Management

### What Convocore Manages

<CardGroup cols={2}>
  <Card title="Server Uptime" icon="check">
    99.9% uptime guarantee with monitoring
  </Card>

  <Card title="Software Updates" icon="arrow-up">
    Automatic security and feature updates
  </Card>

  <Card title="Security" icon="shield">
    Firewall, fail2ban, and intrusion detection
  </Card>

  <Card title="Scaling" icon="chart-line">
    Auto-scaling for high call volumes
  </Card>

  <Card title="Monitoring" icon="eye">
    24/7 monitoring and alerting
  </Card>

  <Card title="Backups" icon="database">
    Regular configuration backups
  </Card>
</CardGroup>

### What You Don't Need to Worry About

* ❌ Server installation or configuration
* ❌ Software updates or patches
* ❌ SSL certificates
* ❌ Firewall configuration
* ❌ Network troubleshooting
* ❌ Capacity planning
* ❌ Security hardening
* ❌ Log management

## Performance & Reliability

### Call Quality

The SIP server is optimized for:

* **Low Latency**: \< 100ms processing time
* **High Quality**: 8kHz audio, clear voice reproduction
* **Reliability**: 99.9% uptime SLA
* **Concurrent Calls**: Supports hundreds of simultaneous calls

### Geographic Distribution

* **Primary Server**: 134.209.121.168
* **Location**: Optimally located for minimal latency
* **Network**: High-speed, redundant connections
* **CDN**: Audio optimized for global delivery

## Monitoring Your Calls

### Call Logs

All calls through the SIP server are logged in your Convocore dashboard:

* View in **Conversations** tab
* Filter by phone number
* See call duration
* Access transcripts
* Review AI agent performance

### Metrics Available

Track these metrics for your SIP trunk:

* Total calls received
* Average call duration
* Success rate
* Failed call reasons
* Peak calling times
* Geographic distribution of callers

## Common Questions

<AccordionGroup>
  <Accordion title="Can I use my own SIP server?">
    No, all SIP trunking uses Convocore's managed server at 134.209.121.168.
    This ensures optimal performance, reliability, and integration with AI
    agents.
  </Accordion>

  <Accordion title="What happens if the server goes down?">
    The server has 99.9% uptime with redundancy. In the rare event of an issue,
    calls will fail at the server level. We have 24/7 monitoring and automatic
    failover systems.
  </Accordion>

  <Accordion title="How many concurrent calls can it handle?">
    The server auto-scales to handle your call volume. Most customers never need
    to worry about capacity. Enterprise customers can contact us for dedicated
    resources.
  </Accordion>

  <Accordion title="Can I see server logs?">
    You can view call logs and metrics in your Convocore dashboard. Low-level
    server logs are managed by our operations team for security and performance
    monitoring.
  </Accordion>

  <Accordion title="Is the connection secure?">
    Yes. While SIP traditionally uses UDP (unencrypted), the connection between
    the SIP server and Convocore's AI infrastructure uses encrypted WebSocket
    (WSS). Your conversation data is protected.
  </Accordion>

  <Accordion title="What if I need a different server location?">
    For enterprise customers with specific latency or compliance requirements,
    we can discuss dedicated server deployments. Contact our sales team.
  </Accordion>
</AccordionGroup>

## Troubleshooting

### If Calls Aren't Connecting

<Steps>
  <Step title="Verify SIP Provider Configuration">
    Ensure your SIP provider is routing to: - IP: `134.209.121.168` - Port:
    `5060` - Protocol: UDP
  </Step>

  <Step title="Check Phone Number Configuration">
    Verify in Convocore dashboard: - Phone number is added - Number is in E.164
    format - Agent is assigned - Agent is voice-enabled
  </Step>

  <Step title="Test SIP Provider">
    Contact your SIP provider to ensure: - The number is active - Call
    forwarding is enabled - No IP restrictions blocking our server - SIP trunk
    is registered
  </Step>

  <Step title="Check Conversation Logs">
    Look in your Convocore dashboard for error messages or failed call attempts
  </Step>
</Steps>

### If Audio Quality is Poor

Poor audio quality usually indicates:

* Network issues with your SIP provider
* Incompatible codec negotiation
* Bandwidth limitations
* Packet loss between provider and our server

**Solutions:**

1. Ask your provider to use PCMU or PCMA codec
2. Check for network congestion
3. Verify your provider's audio quality settings
4. Contact Convocore support with call examples

## Advanced Topics

### Call Detail Records (CDR)

Every call generates a CDR that includes:

* Call start/end time
* Duration
* Source phone number
* Destination (agent ID)
* Call status (answered, failed, etc.)
* Codec used

Access CDRs through your Convocore dashboard or API.

### Audio Format Details

**Standard Call Format:**

```
Sample Rate: 8000 Hz
Bit Depth: 16-bit
Channels: Mono (1 channel)
Codec: PCMU/PCMA
Bitrate: 64 kbps
```

This format is optimized for:

* Low latency voice communication
* Compatibility with telephony standards
* Efficient AI processing
* Clear speech recognition

### Load Balancing

The SIP server uses intelligent load balancing:

* Round-robin for incoming calls
* Health checks for backend services
* Automatic failover
* Geographic routing optimization

## Next Steps

Now that you understand how the SIP server works, you're ready to:

<CardGroup cols={2}>
  <Card title="Configure Your Provider" icon="plug" href="/voice/sip-trunking/adding-provider">
    Set up your SIP trunk provider
  </Card>

  <Card title="Add Phone Numbers" icon="hashtag" href="/voice/sip-trunking/phone-numbers">
    Configure phone numbers and agents
  </Card>
</CardGroup>

## Support

Need help with the SIP server?

* 📧 **Email:** [support@convocore.ai](mailto:support@convocore.ai)
* 💬 **Discord:** [Join our community](https://discord.gg/XrJtBsQQRN)
* 📊 **Status:** [Check server status](https://status.convocore.ai)

<Tip>
  **Remember**: You don't need to manage the server yourself. Just configure
  your SIP provider to route to `134.209.121.168:5060` and Convocore handles the
  rest!
</Tip>