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SIP Integration

SIP (Session Initiation Protocol) integration allows Crew to connect directly to your existing phone infrastructure, PBX systems, or SIP providers.

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Overview

SIP integration is ideal for:
  • Enterprises with existing PBX systems
  • Organizations using custom SIP providers
  • High-volume deployments requiring direct connectivity
  • Scenarios requiring specific codec or routing control

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Prerequisites

Before setting up SIP:
  1. SIP trunk provider or PBX with SIP support
  2. Static IP address(es) or FQDN for Crew
  3. Network firewall access for SIP/RTP traffic
  4. Codec compatibility (G.711, Opus)

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SIP Trunk Configuration

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Crew SIP Endpoint

Configure your SIP trunk to route calls to Crew:
SettingValue
SIP URIsip:{crew_id}@sip.usecrew.ai
TransportUDP, TCP, or TLS
Port5060 (UDP/TCP) or 5061 (TLS)
CodecG.711 μ-law, G.711 A-law, Opus

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Authentication

Crew supports multiple SIP authentication methods: IP Allowlisting: 
{
  "sip": {
    "auth": {
      "type": "ip_allowlist",
      "allowed_ips": [
        "203.0.113.10",
        "203.0.113.11"
      ]
    }
  }
}
Digest Authentication: 
{
  "sip": {
    "auth": {
      "type": "digest",
      "username": "crew_user",
      "password": "secure_password"
    }
  }
}

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Outbound Configuration

For outbound calls via your SIP trunk: 
{
  "sip": {
    "outbound": {
      "uri": "sip:your-trunk.provider.com",
      "transport": "tls",
      "auth": {
        "username": "your_username",
        "password": "your_password"
      },
      "caller_id": "+14155551234"
    }
  }
}

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Call Routing

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Inbound Routing

Map inbound SIP calls to Crew agents: 
{
  "sip": {
    "inbound_routes": [
      {
        "match": {
          "to_user": "main"
        },
        "agent_id": "agent_sarah"
      },
      {
        "match": {
          "to_user": "support"
        },
        "agent_id": "agent_marcus"
      }
    ]
  }
}

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DID Routing

Route based on called number (DID): 
{
  "sip": {
    "inbound_routes": [
      {
        "match": {
          "to_uri": "+14155551111"
        },
        "agent_id": "agent_sales"
      },
      {
        "match": {
          "to_uri": "+14155552222"
        },
        "agent_id": "agent_support"
      }
    ]
  }
}

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Network Configuration

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Firewall Rules

Allow the following traffic:
ProtocolPortDirectionPurpose
UDP5060Inbound/OutboundSIP signaling
TCP5060Inbound/OutboundSIP signaling
TLS5061Inbound/OutboundSecure SIP
UDP10000-20000Inbound/OutboundRTP media

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Crew IP Ranges

For outbound calls from Crew, allow these IP ranges: 
203.0.113.0/24  (example - actual IPs provided during setup)
198.51.100.0/24 (example - actual IPs provided during setup)
Contact support@usecrew.ai for current IP ranges.

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Codec Configuration

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Supported Codecs

CodecSample RateBitrateQuality
G.711 μ-law8kHz64 kbpsStandard
G.711 A-law8kHz64 kbpsStandard
Opus8-48kHz6-510 kbpsHigh

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Codec Priority

Configure codec preference: 
{
  "sip": {
    "codecs": [
      {
        "name": "opus",
        "priority": 1,
        "sample_rate": 16000
      },
      {
        "name": "PCMU",
        "priority": 2
      },
      {
        "name": "PCMA",
        "priority": 3
      }
    ]
  }
}

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PBX Integration

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Asterisk

Example Asterisk trunk configuration: 
; sip.conf
[crew-trunk]
type=friend
host=sip.usecrew.ai
username=your_crew_id
secret=your_sip_password
fromdomain=sip.usecrew.ai
insecure=port,invite
qualify=yes
context=from-crew

; extensions.conf
[from-external]
exten => _X.,1,Dial(SIP/${EXTEN}@crew-trunk)

[from-crew]
exten => _X.,1,Dial(SIP/${EXTEN})

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FreePBX

Configure via FreePBX web interface:
  1. Connectivity → Trunks → Add SIP Trunk
  2. Set outbound caller ID
  3. Configure PEER details with Crew SIP endpoint
  4. Create outbound route pointing to trunk

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Cisco Unified CM

Configure SIP trunk in CUCM:
  1. Device → Trunk → Add New
  2. Select SIP Trunk type
  3. Configure destination as sip.usecrew.ai
  4. Set up route pattern for Crew destinations

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SIP Headers

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Custom Headers

Pass custom data via SIP headers: 
{
  "sip": {
    "custom_headers": {
      "inbound": [
        {
          "header": "X-Customer-ID",
          "variable": "customer_id"
        }
      ],
      "outbound": [
        {
          "header": "X-Campaign-ID",
          "value": "{{campaign_id}}"
        }
      ]
    }
  }
}

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RPID/PAI

Configure Remote Party ID or P-Asserted-Identity: 
{
  "sip": {
    "caller_id": {
      "source": "pai",
      "fallback": "rpid"
    }
  }
}

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Monitoring

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SIP Registration Status

Check SIP trunk status: 
curl https://api.usecrew.ai/v1/sip/status \
  -H "Authorization: Bearer YOUR_API_KEY"

{
  "status": "registered",
  "registrar": "sip.usecrew.ai",
  "last_registration": "2024-01-15T10:00:00Z",
  "expires": "2024-01-15T10:05:00Z"
}

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Call Quality Metrics

Monitor SIP call quality: 
{
  "call_id": "call_abc123",
  "quality": {
    "mos": 4.2,
    "jitter_ms": 12,
    "packet_loss_percent": 0.1,
    "rtt_ms": 45
  }
}

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Troubleshooting

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Registration Failures

  1. Verify credentials are correct
  2. Check firewall allows SIP traffic
  3. Ensure DNS resolves properly
  4. Check for NAT traversal issues

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One-Way Audio

  1. Verify RTP ports are open
  2. Check for NAT/firewall blocking
  3. Ensure symmetric RTP if required
  4. Verify codec negotiation

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Call Quality Issues

  1. Check network latency and jitter
  2. Verify sufficient bandwidth
  3. Test codec compatibility
  4. Review QoS settings

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Best Practices

Encrypt SIP signaling with TLS for security.
Use SRTP to encrypt voice traffic when possible.
Prioritize SIP/RTP traffic on your network.
Track MOS scores and address degradation promptly.
Configure backup SIP trunks for redundancy.

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Next Steps