When you have two Ubuntu systems in different locations, you can test voice traffic using several approaches. Below are three main methods:

sudo apt-get update
sudo apt-get install asterisk

Configure Extensions:Edit Asterisk configuration files (usually sip.conf and extensions.conf) to create SIP extensions for both systems.

sudo apt-get install linphone

Test Calls:Register the SIP client with the Asterisk server and initiate a call.Use tools such as Wireshark to capture RTP packets, analyzing jitter, delay, and packet loss.

Wireshark:Capture and analyze RTP streams.

RTP Analysis Tools:Tools like rtpdump can help in examining the RTP stream details.

SIPp is a powerful tool to simulate SIP calls and generate voice traffic:

Install SIPp on your Ubuntu system:

sudo apt-get install sipp

SIPp comes with built-in scenarios (e.g., uac.xml) to simulate SIP call flows, including media exchange.

To simulate a basic SIP call:

sipp -sf uac.xml [remote_IP] -s 100

Adjust the parameters as needed for your test.

VoIP Monitoring Tools:Consider using tools such as VoIPmonitor for in-depth analysis of latency, jitter, and packet loss.

Direct Softphone Testing:Run softphones (like Linphone or Ekiga) on both systems for simple call tests, then use system-level tools (e.g., netstat or tc with netem) to simulate various network conditions.

Asterisk + Linphone:Ideal for setting up a full SIP environment to test manual call setups and monitor media traffic.

SIPp:Useful for automated testing and performance metrics.

Monitoring Tools:Wireshark and VoIPmonitor help you analyze the quality of RTP streams and network performance.

Choose the approach that best suits your testing needs, whether you prefer a hands-on manual test or automated performance testing.

Feel free to ask for further details or configuration examples!