Tuesday, 7 October 2014

Build a test bed with FreeSWITCH and PJSUA (with SILK)

Build and run FreeSWITCH

This is based on debian wheezy, and uses master FreeSWITCH. You can switch to stable when cloning.
In my experience this is the quickest way to get to a working, vanilla setup that you can use to automate tests with PJSUA (the PJSIP command-line client).

mkdir git
cd git
# To get master
git clone https://stash.freeswitch.org/scm/fs/freeswitch.git
cd git/freeswitch
./bootstrap.sh -j
# If changes are needed to the list of modules to be compiled
vi modules.conf
./configure --enable-core-pgsql-support
make
sudo make install
# Launch FS (in foreground)
sudo /usr/bin/freeswitch

You can find here the process for building it from source recommended by the FreeSWITCH team.

In /etc/freeswitch/vars.xml you may want to change the default password in:
cmd="set" data="default_password=XXXXXXXX"

If you're enabling SILK, add it to the list in modules.conf.xml:
load module="mod_silk"

and add it to global_codec_prefs and outbound_codec_prefs inside vars.xml, e.g.:
cmd="set" data="global_codec_prefs=SILK@24000h@20i,SILK@16000h@20i,SILK@8000h@20i,speex@16000h@20i,speex@8000h@20i,PCMU,PCMA"
cmd="set" data="outbound_codec_prefs=SILK@24000h@20i,SILK@16000h@20i,SILK@8000h@20i,speex@16000h@20i,speex@8000h@20i,PCMU,PCMA"


Build and run PJSUA

Note that you need to retrieve a copy of the SILK SDK, once available from skype's developer site (and put it in ~/silk-1.0.9.zip):

mkdir pjsip && cd pjsip
svn co http://svn.pjsip.org/repos/pjproject/trunk/@4806 pjproject
cp ~/silk-1.0.9.zip .
unzip silk-1.0.9.zip && cd SILK_SDK_SRC_FLP_v1.0.9 && make clean all
cd ../pjproject
./configure --with-silk=pjsip/SILK_SDK_SRC_FLP_v1.0.9
make dep && make

Now create a configuration file for PJSUA in pjproject dir, e.g. pjsua_test.cfg:

--id sip:1001@192.168.0.10
--registrar sip:192.168.0.10
--realm 192.168.0.10
--username 1001
--password THEPASSWORD
--local-port 5066
# SRTP - 0:disabled, 1:optional, 2:mandatory (def:0)
--use-srtp 0
# SRTP require secure SIP? 0:no, 1:tls, 1:sips (def:1)
--srtp-secure=0
# Disable the sound device. Calls will behave normally, except that no audio will be transmitted or played locally.
--null-audio
# Automatically stream the WAV file to incoming calls.
--auto-play
--play-file tests/pjsua/wavs/input.8.wav

where 192.168.0.10 is FreeSWITCH's listening IP address.
PJSUA can be local (i.e. run on the same host as FreeSWITCH), or remote (with obvious network reachability constraints).

Run PJSUA from pjproject with:
./pjsip-apps/bin/pjsua-x86_64-unknown-linux-gnu --config-file=pjsua_test.cfg

and let's the tests begin!

You may want to increase FreeSWITCH verbosity by enabling SIP traces and debug messages in fs_cli with:

sofia global siptrace on
console loglevel debug
It's also easy to use the same configuration and have PJSUA as one of the parties (caller or callee) and have a softphone on the other side, like Blink or X-Lite.
The advantage of PJSUA is that you can easily automate its execution and result parsing, for example with its python module.

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