Voice Gateway Setup

The voice gateway (cdma-voice-gw) is a separate process that bridges CDMA voice calls to SIP/PSTN networks. It handles SIP signaling, RTP media, and EVRC↔G.711 transcoding.

Architecture

CDMA Handset
    ↕ air interface (EVRC)
BTS / BSC / MSC media + policy
    ↕ A1 call control + A2p RTP bearer on the cellular side
cdma-voice-gw
    ↕ SIP/RTP (G.711)
SIP Trunk / PSTN

The cellular side separates call control from bearer media: MSC owns call policy, media orchestration, gateway control, and preemption over A1/A2p, while BSC executes the radio leg. The gateway remains the external SIP/RTP bridge for non-local calls.

MSC Configuration

Enable the gateway in config/msc.json:

{
  "voice": {
    "gateway": {
      "enabled": true,
      "endpoint": "http://127.0.0.1:17015",
      "fallback_to_wav": true
    }
  }
}

Gateway Configuration

The gateway uses its own JSON config file. A complete example is at config/voice-gw.example.json.

gRPC

{
  "grpc": {
    "listen_addr": "127.0.0.1:17015"
  }
}

SIP

{
  "sip": {
    "listen_addr": "192.168.1.50:5060",
    "transport": "udp",
    "request_uri_template": "sip:{called}@trunk.example.com:5060",
    "from_domain": "bts.local",
    "caller_id_override": null,
    "user_agent": "1XBTS-VoiceGW/0.1",
    "keepalive_interval_secs": 0,
    "auth": {
      "username": "mytrunk",
      "password_env": "SIP_PASSWORD"
    },
    "registration": {
      "enabled": false,
      "registrar_uri": "sip:trunk.example.com",
      "expires_secs": 300
    }
  }
}

RTP

{
  "rtp": {
    "listen_addr": "0.0.0.0",
    "port_range": [17100, 17200],
    "advertise_addr": null,
    "preferred_codecs": ["PCMU", "PCMA"]
  }
}

NAT

{
  "nat": {
    "mode": "disabled",
    "stun_server": "stun.l.google.com:19302"
  }
}

Calls

{
  "calls": {
    "max_concurrent_calls": 32,
    "setup_timeout_ms": 30000,
    "ringing_timeout_ms": 120000,
    "media_idle_timeout_ms": 30000
  }
}

Jitter Buffer

{
  "jitter_buffer_ms": 60
}

Adaptive buffer (40-80 ms) for inbound RTP reordering. Inserts EVRC silence frames when packets are late.

Running the Gateway

# Set SIP password
export SIP_PASSWORD="mysecret"

# Start the gateway
cargo run --release -p cdma-voice-gw -- --config config/voice-gw.example.json

The gateway listens on the configured gRPC port. The MSC runtime connects automatically when gateway.enabled is true.

SIP Trunk Providers

The gateway works with any standard SIP trunk. Tested configurations:

FreeSWITCH Example

{
  "sip": {
    "request_uri_template": "sip:{called}@192.168.1.200:5060",
    "from_domain": "192.168.1.50",
    "auth": null,
    "registration": { "enabled": false }
  }
}

Flowroute Example

{
  "sip": {
    "request_uri_template": "sip:{called}@us-west-wa.sip.flowroute.com",
    "from_domain": "sip.flowroute.com",
    "auth": {
      "username": "your-tech-prefix",
      "password_env": "FLOWROUTE_PASSWORD"
    },
    "registration": {
      "enabled": true,
      "registrar_uri": "sip:sip.flowroute.com",
      "expires_secs": 300
    }
  }
}

Transcoding

The gateway transcodes between EVRC (air interface) and G.711 (SIP/RTP):

Media frames on the cellular bearer are exchanged as raw EVRC codec payloads (without MuxPDU headers). The BSC adds or strips the MuxPDU MM=0 header for the Abis and air-interface side, and the MSC forwards non-local call media between A2p and the gateway. A2p uses RTP/UDP per circuit and the EVRC RTP payload format from RFC 3558. Erasure frames with rate_bps=0 are skipped in the gateway media path.

Troubleshooting