See the Pen CppComet video chat example by Trapenok Victor (@Levhav) on CodePen.
====== Conception of API video chats ====== {{ :comet:video:video-api.gif |Comet Video API}} * For work of video of chats of comets a server uses possibilities of FreeSwitch simply co-operating with him on API. * Comets a server is able to distribute video and audio calls on the great number of servers with FreeSwitch. * Work with FreeSwitch fully hides after API comets of server. ====== CometQL API ====== It is added the table to CometQL of conference that contains information about active conferences. To create a new conference and add a member to it, you must complete the query
INSERT INTO conference (name, user_id, caller_id, message, profile)VALUES('1', 2, 2, "", "video");
There are parameters:
* name - the name of conference (consists only of numbers)
* of user _ id is an identifier of the user
* of caller _ id added to the conference is an identifier of that user a that initiates call (not required parameter)
* of message is an arbitrary report
* of profile is the mode of call the variants " of audio" are possible, " video" (In open source of version and on demand in support in a saas version it is possible to create the modes of video of conferences that to add thread-marks, show not simply talking the moment participant of conference in given, and all at once or to influence the rules of show of participants to the conference on a general canvas. All possibilities are enumerated in documentation on [[https://freeswitch.org/confluence/display/FREESWITCH/mod_conference|mod _ conference]])
To add another participant it is necessary to execute the same query only to pass new user _ id and old name. Accordingly creation of conference from 5 participants will demand 5 queries of insertion in the table of conference
An example can be viewed in the file [[https://github.com/CppComet/video-chat-example|call.php]]
Selecting data from the table conference
select * from `conference` where name = "101";
Deleting data from the table conference
delete from `conference` where name = "101" and user_id = 29130;
select * from `conference` where name = "101";
It does not mean that all these people are still in this conference.
To know who presently speaks it is necessary to inquire to the table `conference_members`
select * from `conference_members`
Presently the table 'of conference _ members' is accessible read-only and contains such fields
* of name - the name of conference
* of user _ id is an identifier of user
* of join _ time - the amount of seconds passing from the moment of connecting of user
* of last _ talking - the amount of seconds passing from that moment when he that talked that in a conference.
====== JavaScript API ======
In js it is necessary to connect a file [[https://github.com/CppComet/comet-server/tree/master/api|cometVideoApi.js]] and library [[https://github.com/CppComet/comet-server/blob/alfa/api/jssip.js|JsSIP v2.0.6]]
whereupon for treatment of events of incoming call, connecting and disconnecting it is necessary to execute a next code
// Initiate activation cometVideoApi
cometVideoApi.start({
// Callback caused before the beginning of connecting for a call
// Assumed that in him tuning will be set for the nearest call
// Such as well as parameters of audio _ remote, video _ local, video _ remote and maybe yet some.
// And then the function of cometVideoApi will be caused.acceptCall (event)
// And if will not be caused that means we did not take a handset.
onCall:function(callEvent)
{
$(".status").html("Incoming call, press answer.");
if(!confirm("Answer a call?"))
{
// It was decided not to answer the call
$(".status").html("It was decided not to answer the call.");
$(".StartCallBtn").show();
return
}
// Take the handset if you want
cometVideoApi.acceptCall({
// Type of call 'audio' | 'video'
type:'video',
// Specify the target element for the video stream from the interlocutor
video_remote: $("#video_remote")[0],
// Specify the target element for the audio stream from the interlocutor
audio_remote: $("#audio_remote")[0],
// Specify the target element for the video stream from me (my own image from the camera)
video_local: $("#video_local")[0],
})
$(".status").html("Waiting for a response from other conference participants ");
},
/**
* Callback to completion callback
* @param {object} event
* {
* action: "", // Event name
* status: "", // Call completion reason
* callInfo: {}, // Information about the call
* time: 1000 // Time duration of the call
* type: "audio" // Type of call * }
*/
onCallEnd:function(event)
{
$(".root").removeClass("incall")
$(".StartCallBtn").show();
$(".status").html("Call is completed ");
log("onCallEnd", event)
},
/**
* Callback of raising a handset by interlocutor
* @param {object} event
* {
* action:"accept", // Event name
* callInfo:{}, // Information about the call
* type:"audio" // The type of call that is selected by the interlocutor
* }
*
* Callback is called only once for the first responding interlocutor
*/
onCallAccept: function(event)
{
$(".status").html ("A response is received from another participant.");
log("onAccept", event)
},
/**
* Callback when I and my interlocutor were connected to the server
* @param {object} event
* {
* action:"start", // Event name
* callInfo:{}, // Information about the call
* type:"audio" // The type of call
* }
*/
onCallStart: function(event)
{
$(".status").html("Connecting to the mediaserver is executed. Conversation began.");
$(".root").addClass("incall")
log("onCallStart", event)
},
onOut: function(event)
{
// Exit of participant from a conference
$(".status").html ("One of interlocutors abandoned a conference");
log("onOut", event)
},
onIn: function(event)
{
// Log in of participant to the conference
$(".status").html("Another person joined the conference.");
log("onIn", event)
},
})
====== Set up a backend for video calls ======
[freeswitch]
ip = 0.0.0.0
thread_num = 3
statistics = 10
port = 84
[sip]
port = 7443
host = ecort-n1.comet.su;
pipesalt = pipeSecretSalt
===== Assembling of FreeSwitch from source codes with the necessary modules =====
It is needed to collect FreeSwitch on instruction [[https://freeswitch.org/confluence/display/FREESWITCH/Debian+8+Jessie#Debian8Jessie-BuildingFromSource|FreeSwitch Building from source]]
Then to plug in assembling and in an autoload next modules
* [[https://freeswitch.org/confluence/display/FREESWITCH/RTMP+Configuration+Files|mod_rtmp]] - will allow to conduct translations of video of calls and conferences
* [[https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl|mod_xml_curl]] - FreeSwitch will allow to use the same these authorizing a FreeSwitch server that present in comets server in the table of users _ auth * mod_av
* mod_sndfile
* mod_shell_stream
* mod_ilbc
* mod_h26x
* mod_siren
* mod_isac
Final script for assembling with the necessary modules
wget -O - https://files.freeswitch.org/repo/deb/debian/freeswitch_archive_g0.pub | apt-key add -
echo "deb http://files.freeswitch.org/repo/deb/freeswitch-1.6/ jessie main" > /etc/apt/sources.list.d/freeswitch.list
apt-get update
apt-get install -y --force-yes freeswitch-video-deps-most
# because we're in a branch that will go through many rebases it's
# better to set this one, or you'll get CONFLICTS when pulling (update)
git config --global pull.rebase true
# then let's get the source. Use the -b flag to get a specific branch
cd /usr/src/
git clone https://freeswitch.org/stash/scm/fs/freeswitch.git -bv1.6 freeswitch
cd freeswitch
./bootstrap.sh -j
sed -e "s/#codecs\/mod_isac/codecs\/mod_isac/g" /usr/src/freeswitch/modules.conf | sed -e "s/#applications\/mod_av/applications\/mod_av/g" | sed -e "s/#endpoints\/mod_rtmp/endpoints\/mod_rtmp/g" | sed -e "s/#xml_int\/mod_xml_curl/xml_int\/mod_xml_curl/g" | sed -e "s/#applications\/mod_av/applications\/mod_av/g"| sed -e "s/#formats\/mod_sndfile/formats\/mod_sndfile/g" | sed -e "s/#formats\/mod_shell_stream/formats\/mod_shell_stream/g" | sed -e "s/#codecs\/mod_ilbc/codecs\/mod_ilbc/g"| sed -e "s/#codecs\/mod_h26x/codecs\/mod_h26x/g" | sed -e "s/#codecs\/mod_siren/codecs\/mod_siren/g"| sed -e "s/#codecs\/mod_isac/codecs\/mod_isac/g" > /usr/src/freeswitch/modules.tmp
cat /usr/src/freeswitch/modules.tmp > /usr/src/freeswitch/modules.conf
rm -rf /usr/src/freeswitch/modules.tmp
./configure
make
make install
# create user 'freeswitch'
# add it to group 'freeswitch'
# change owner and group of the freeswitch installation
cd /usr/local
groupadd freeswitch
adduser --quiet --system --home /usr/local/freeswitch --gecos "FreeSWITCH open source softswitch" --ingroup freeswitch freeswitch --disabled-password
chown -R freeswitch:freeswitch /usr/local/freeswitch/
chmod -R ug=rwX,o= /usr/local/freeswitch/
chmod -R u=rwx,g=rx /usr/local/freeswitch/bin/*
sed -e "s/EnvironmentFile=-\/etc\/default\/freeswitch/EnvironmentFile=-\/etc\/freeswitch/g" /usr/src/freeswitch/debian/freeswitch-systemd.freeswitch.service | sed -e "s/PIDFile=\/run\/freeswitch\/freeswitch.pid/PIDFile=\/usr\/local\/freeswitch\/run\/freeswitch.pid/g" > /etc/systemd/system/freeswitch.service
cp /usr/local/freeswitch/bin/freeswitch /usr/bin/freeswitch
mkdir /run/freeswitch
systemctl daemon-reload
systemctl enable freeswitch
It is then necessary to execute instructions [[https://freeswitch.org/confluence/display/FREESWITCH/Debian+Post-Install+Tasks|Debian+Post-Install+Tasks]]
===== Plugging of the modules in an autoload =====
To plugging the startup modules, you need to make sure that the file /usr/local/freeswitch/conf/autoload_configs/modules.conf.xml has the following lines
==== Tuning xml_curl.conf on freeswitch ====
In the /usr/local/freeswitch/conf/autoload_configs/xml_curl.conf.xml file, you need to enter the port and the address to connect to the comet server with the requisites that we specified in the section
[freeswitch]
This tuning is responsible for that users that is authorized on comets server could from the same data of authorizing connected to freeswitch for the feasance of call.
Here are the details of the module work [[https://freeswitch.org/confluence/display/FREESWITCH/mod_xml_curl+PHP+example|mod_xml_curl]]
==== Tuning conference.conf.xml on freeswitch ====
For tuning of different profiles of work of video of conferences it is necessary to edit a file /usr/local/freeswitch/conf/autoload_configs/conference.conf.xml
Documentation is on [[https://freeswitch.org/confluence/display/FREESWITCH/mod_conference|mod_conference]]
the default settings are suitable.
If you want to broadcast a conference of viewers on rtmp then in the settings of the conference profile you can enter the command:
With the indication of your nginx with nginx-rtmp-module
==== Tuning dialplan/default.xml on freeswitch ====
The /usr/local/freeswitch/conf/dialplan/default.xml file contains dialing commands. He used to determine what to do when calling for real numbers.
In the section you need to add an extension to the video conferencing set
==== Tuning vars.xml on freeswitch ====
Useful in vars.xml to set a proof password in a line, he will be used comets by a server for api calls.
It is needed to set the variable domain=FSdefaultDomain
==== Tuning event_socket.conf on freeswitch ====
In event_socket.conf, configure which port and with what password freeswitch will allow to connect to execute api commands.
==== Tuning https on freeswitch ====
Usually after setting of freeswitch the self-signed certificate lies in the folder of /usr/local/freeswitch/certs/wss.pem. For work of video and audio chats of them it is necessary to substitute by valid ones because differently from a https web-site will not turn out connected to freeswitch on websockets for an exchange by sip commands with freeswitch
For the receipt of certificates it comfortably to use Let's Encrypt. I use [[https://habrahabr.ru/post/318952/|this instruction]]
And as post - hook for updating of certificate such command:
cat /etc/letsencrypt/live/fs-n3.elevenow.com/fullchain.pem > /usr/local/freeswitch/certs/wss.pem && cat /etc/letsencrypt/live/fs-n3.elevenow.com/privkey.pem >> /usr/local/freeswitch/certs/wss.pem && cat /etc/letsencrypt/live/fs-n3.elevenow.com/fullchain.pem > /usr/local/freeswitch/certs/dtls-srtp.pem && cat /etc/letsencrypt/live/fs-n3.elevenow.com/privkey.pem >> /usr/local/freeswitch/certs/dtls-srtp.pem && systemctl stop freeswitch && rm -rf /var/lib/freeswitch/db && systemctl start freeswitch
====== Tuning broadcasts in nginx =====
For broadcasts, you need to compile nginx with the nginx-rtmp-module module here: [[https://www.dmosk.ru/instruktions.php?object=nginx-rtmp#build|weak instruction]]
Here is the example of configuration nginx
worker_processes 1;
rtmp {
server {
live on;
listen 1935;
chunk_size 512;
buflen 1s;
idle_streams off;
application cam1 {
live on;
record off;
hls on;
hls_path /tmp/hls;
hls_fragment 2s;
hls_playlist_length 20s;
hls_type live;
#hls_continuous on;
hls_fragment_naming sequential;
hls_nested on;
dash on;
dash_path /tmp/dash;
dash_fragment 2s;
dash_playlist_length 10s;
dash_nested on;
}
}
}
server {
listen 80 default_server;
listen [::]:80 default_server;
root /var/www/html;
index index.html index.htm index.nginx-debian.html;
server_name _;
include acme;
location /dash {
root /tmp;
add_header Access-Control-Allow-Origin *;
add_header Cache-Control no-cache;
}
location / {
add_header Cache-Control no-cache;
add_header Access-Control-Allow-Origin *;
root /tmp;
}
}
server {
listen 0.0.0.0:443;
server_name comet-server.com;
ssl on;
ssl_certificate /etc/letsencrypt/live/stream-n1.example.com/fullchain.pem;
ssl_certificate_key /etc/letsencrypt/live/stream-n1.example.com/privkey.pem;
ssl_session_timeout 70m;
keepalive_disable none;
lingering_close always;
send_timeout 3600s;
client_max_body_size 100m;
include acme;
location /dash {
root /tmp;
add_header Access-Control-Allow-Origin *;
add_header Cache-Control no-cache;
}
location / {
add_header Cache-Control no-cache;
add_header Access-Control-Allow-Origin *;
root /tmp;
}
}
These articles can be useful
* https://github.com/arut/nginx-rtmp-module/wiki/Directives
* https://habrahabr.ru/post/162237/#workers