VoIP Infrastrastructure & SIP Trunking Integration

Packet 111 (INVITE): The Call Request

• • The FreePBX server sends a secure request with the correct account credentials to VoIP.ms to start the outbound call.

Packet 116 (100 Trying): The Network Acknowledgment

• • VoIP.ms replies to the FreePBX server to say, "We got your message and are working on routing the call right now."

Packet 281 (183 Session Progress): The Ringing Signal

• • VoIP.ms opens up the audio channel early so I can hear my cell phone ringing on my Grandstream desk phone.

Packet 374 (200 OK): Start of Call

• • The exact moment my cell phone answers the call, opening up the live, two-way voice stream back to my Grandstream phone.

Packet 13366 (BYE): End of Call

• • VoIP.ms sends a message to my FreePBX server to tear down the connection the exact second the cellular phone hangs up.

Packet 13367 (200 OK): Call Closed

• • My FreePBX server sends a final confirmation back to VoIP.ms showing the call successfully finished and the network line is completely cleared out.

G.711 Call Performance Stats

• Audio Codec: G.711 PCMU

• Voice Payload Size: 160 bytes

• Network Bandwidth: 80 kbps 

• Average Jitter: 0.16 ms (<20 ms)

• Maximum Jitter: 0.50 ms (<30 ms)

• Total Packets: 1,581 packets

• Packet Loss: 0% (<1%)