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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.

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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
Important: The SIP server is fully managed by Convocore. You don’t need to install, configure, or maintain anything.

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Server Details

Your SIP provider should route calls to: 
IP Address: 161.35.79.143
Port: 5060
Protocol: UDP (SIP)

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How It Works

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Call Flow

When someone calls your phone number:

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Step-by-Step Process

1

Call Arrives

Your SIP provider receives an incoming call and routes it to 161.35.79.143:5060
2

Number Lookup

The SIP server looks up the phone number in Convocore’s database to find the assigned agent
3

Agent Connection

The server establishes a WebSocket connection to the assigned AI agent
4

Audio Bridge

Audio flows bidirectionally:
  • Caller’s voice → SIP Server → AI Agent
  • AI Agent’s response → SIP Server → Caller
5

Real-Time Processing

The AI agent processes speech, generates responses, and speaks back through the bridge
6

Call Termination

When either party hangs up, the connection is cleanly terminated

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Technical Architecture

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Components

SIP Handler

Manages SIP signaling (call setup, teardown, etc.)

Audio Engine

Processes PCM16 8kHz mono audio streams

Call Router

Routes calls to the correct AI agent based on phone number

WebSocket Bridge

Connects to Convocore’s AI agent infrastructure

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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

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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

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Network Requirements

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For Your SIP Provider

Your SIP provider needs to:

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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.

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Server Management

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What Convocore Manages

Server Uptime

99.9% uptime guarantee with monitoring

Software Updates

Automatic security and feature updates

Security

Firewall, fail2ban, and intrusion detection

Scaling

Auto-scaling for high call volumes

Monitoring

24/7 monitoring and alerting

Backups

Regular configuration backups

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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

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Performance & Reliability

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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

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Geographic Distribution

  • Primary Server: 161.35.79.143
  • Location: Optimally located for minimal latency
  • Network: High-speed, redundant connections
  • CDN: Audio optimized for global delivery

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Monitoring Your Calls

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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

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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

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Common Questions

No, all SIP trunking uses Convocore’s managed server at 161.35.79.143. This ensures optimal performance, reliability, and integration with AI agents.
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.
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.
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.
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.
For enterprise customers with specific latency or compliance requirements, we can discuss dedicated server deployments. Contact our sales team.

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Troubleshooting

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If Calls Aren’t Connecting

1

Verify SIP Provider Configuration

Ensure your SIP provider is routing to:
  • IP: 161.35.79.143
  • Port: 5060
  • Protocol: UDP
2

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
3

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
4

Check Conversation Logs

Look in your Convocore dashboard for error messages or failed call attempts

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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

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Advanced Topics

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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.

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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

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Load Balancing

The SIP server uses intelligent load balancing:
  • Round-robin for incoming calls
  • Health checks for backend services
  • Automatic failover
  • Geographic routing optimization

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Next Steps

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

Configure Your Provider

Set up your SIP trunk provider

Add Phone Numbers

Configure phone numbers and agents

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Support

Need help with the SIP server?
Remember: You don’t need to manage the server yourself. Just configure your SIP provider to route to 161.35.79.143:5060 and Convocore handles the rest!