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Kamailio Voice over IP

Load Testing with SIPp

SIPp (a performance testing tool for SIP) to simulate many concurrent calls leading to many concurrent transcoding sessions.

I recently wrote a post on software-based transcoding limits on common virtualisation hardware.

To do this I needed to make a lot of calls, consistently, so as to generate some pretty graphs and stats.

To do this, I used SIPp (a performance testing tool for SIP) to simulate many concurrent calls leading to many concurrent transcoding sessions.

Installation

I built SIPp on Ubuntu 18.04:

apt-get install make autoconf pkg-config automake libncurses5-dev libpcap*
git clone https://github.com/SIPp/sipp.git
cd sipp/
./build.sh --with-rtpstream 
cp sipp /usr/local/bin/

Next I setup RTPengine and setup Kamailio to use it.

I modified the Kamailio config allow Transcoding, as I talked about in the post on setting up Transcoding in RTPengine with Kamailio.

Now I had a working Kamailio instance with RTPengine that was transcoding.

So the next step becomes testing the transcoding is working, for this I had two SIPp instances, one to make the calls and once to answer them.

Instance 1

Makes calls to the IP of the Kamailio / RTPengine instance, for this I modified the uac_pcap scenario to playback an RTP stream of a PCMA (G.711 a-law) call to the called party (stored in a pcap file), and made it call the Kamailio instance multiple times based on how many concurrent transcoding sessions I wanted:

sipp -m 120 -r 200 -sf uac_pcap.xml rtpenginetranscode.nickvsnetworking.com

Instance 2

Instance 2 acted as a simple SIP UAS, the call came in, it answered and echoed back the RTP stream it received.

sipp -rtp_echo -sf uas.xml

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