Audio Codecs

SimpleSIP supports multiple audio codecs for different quality and bandwidth requirements.

Supported Codecs

G.722 (High Quality)

Sample Rate: 16 kHz (internally), 8 kHz RTP clock
Bit Rate: 64 kbps
Quality: High definition audio
RTP Payload Type: 9
Use Case: Best quality, modern SIP servers

# G.722 is automatically preferred when available
client = SimpleSIPClient("user", "pass", "server.com")
client.connect()
client.make_call("1234")

# Check if G.722 was negotiated
if client.negotiated_codec == "G722":
    print("Using high-quality G.722 codec!")

PCMU (G.711 μ-law)

Sample Rate: 8 kHz
Bit Rate: 64 kbps
Quality: Standard telephony quality
RTP Payload Type: 0
Use Case: Default codec, universal compatibility

# PCMU is the fallback codec
# Most compatible with legacy systems

PCMA (G.711 A-law)

Sample Rate: 8 kHz
Bit Rate: 64 kbps
Quality: Standard telephony quality
RTP Payload Type: 8
Use Case: European/international systems

Codec Negotiation

SimpleSIP automatically negotiates the best available codec:

G.722 - Preferred for high quality
PCMU - Fallback for compatibility
PCMA - Alternative fallback

client = SimpleSIPClient("user", "pass", "server.com")
client.connect()
client.make_call("1234")

# Wait for negotiation
while not client.negotiated_codec:
    time.sleep(0.1)

print(f"Negotiated codec: {client.negotiated_codec}")
print(f"Payload type: {client.negotiated_payload_type}")

Audio Quality Comparison

Codec	Sample Rate	Bit Rate	Quality
G.722	16 kHz	64 kbps	HD Audio
PCMU	8 kHz	64 kbps	Standard
PCMA	8 kHz	64 kbps	Standard

Configuring Server for G.722

To enable G.722 on your SIP server:

Asterisk Configuration

Add to /etc/asterisk/sip.conf or /etc/asterisk/pjsip.conf:

[general]
disallow=all
allow=g722
allow=ulaw
allow=alaw

[1001]  ; Your extension
type=friend
secret=password
host=dynamic
context=default
disallow=all
allow=g722
allow=ulaw

FreeSWITCH Configuration

Add to your dialplan or user configuration:

<action application="set" data="codec_string=G722,PCMU,PCMA"/>

Testing Codec Quality

Use this script to test different codecs:

from simplesip import SimpleSIPClient
import time

def test_codec_quality():
    client = SimpleSIPClient("1001", "password", "server.com")

    try:
        client.connect()
        time.sleep(2)

        print("Making test call...")
        client.make_call("1002")

        # Wait for connection
        while client.call_state.value != 'connected':
            time.sleep(0.1)

        codec = client.negotiated_codec
        sample_rate = client.audio_sample_rate

        print(f"Negotiated Codec: {codec}")
        print(f"Sample Rate: {sample_rate} Hz")
        print(f"Expected Quality: {'HD' if codec == 'G722' else 'Standard'}")

        if codec == "G722":
            print("✅ High quality G.722 active!")
        else:
            print("ℹ️  Using standard quality codec")
            print("   Consider enabling G.722 on your server")

    finally:
        client.disconnect()

if __name__ == "__main__":
    test_codec_quality()

Troubleshooting Codec Issues

No G.722 Support

If G.722 is not being negotiated:

Check server configuration - Ensure G.722 is enabled
Check extension settings - Extension must allow G.722
Network issues - Some NAT devices block G.722

# Force codec preference (server must support it)
# This is done automatically by SimpleSIP

Audio Quality Issues

For poor audio quality:

Check sample rate - Must match codec (8kHz vs 16kHz)
Buffer size - Use 20ms audio chunks (160/320 samples)
Network jitter - Enable jitter buffer on server

# Monitor audio levels
def audio_callback(pcm_data, format_type):
    if format_type == 'pcm':
        import numpy as np
        samples = np.frombuffer(pcm_data, dtype=np.int16)
        rms = np.sqrt(np.mean(samples**2))
        print(f"Audio level: {20 * np.log10(rms):.1f} dB")

client.set_audio_callback(audio_callback, 'pcm')

Best Practices

Use G.722 when possible - Significantly better quality
Match sample rates - 16kHz for G.722, 8kHz for G.711
Monitor codec negotiation - Check what was actually negotiated
Test with different servers - Some have better G.722 support
Consider network bandwidth - All supported codecs use 64kbps

The codec choice significantly impacts audio quality. G.722 provides much clearer, more natural-sounding audio compared to traditional G.711 codecs.