...

Supported platforms and browsers

Supported protocols

• WebRTC
• RTP
• SIP

...

keytool -importcert -keystore /usr/java/jdk1.8.0_181/jre/lib/security/cacerts -storepass changeit -file pbx.crt -alias "pbx"

6. Restart WCS.

Known issues

1. It's impossible to make a SIP call if 'SIP Login' and 'SIP Authentification name' fields are incorrect

Symptoms: SIP call stucks in PENDING state.

Solution: according to the standard, 'SIP Login' and 'SIP Authentification name' should not contain any of unescaped spaces and special symbols and should not be enclosed in angle brackets '<>'.

...

Connection to an existing session

Sometimes it is necessary to connect to already existing session and receive an incoming call. It is usually actual on mobile devices where websocket session is closed automatically when browser goes to background. In this case, only push notifications are available. To keep the sesssion active after disconnection, the keepAlive option should be set while creating the session

sipLogin='Ralf C12441@host.com'
sipAuthenticationName='Ralf C'
sipPassword='demo'
sipVisibleName='null'

and this is allowed

sipLogin='Ralf_C12441'
sipAuthenticationName='Ralf_C'
sipPassword='demo'
sipVisibleName='Ralf C'

2. There are some problems with sound while SIP calls from Edge browser.

Symptoms:

a) The outgoing sound is sometimes abruptly muffled, then it goes normally.

b) The incoming sound is heard only if you speak into the microphone.

Solution:

Switch SILK and G.722 codecs usage off in SIP calls for Edge browser with stripCodecs option:

var outCall = session.createCall({
    callee: $("#callee").val(),
    visibleName: $("#sipLogin").val(),
    localVideoDisplay: localDisplay,
    remoteVideoDisplay: remoteDisplay,
    constraints: constraints,
    receiveAudio: true,
    receiveVideo: false,
    stripCodecs: "silk,g722"
    ...
});

outCall.call();

...

    var connectionOptions = {
        urlServer: url,
        keepAlive: true,
        sipOptions: sipOptions
    };
    ...
    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session, connection){
        ...
    });

In this case, session stays active until the following interval in milliseconds is expired (3600 seconds, or 1 hour by default)

client_timeout=3600000

This interval is periodically checked. The checking priod is set in milliseconds by the following parameter (300 seconds, or 5 minutes by default)

client_timeout_check_interval=300000

A session token should be stored while creating the session

    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session, connection){
        authToken = connection.authToken;
        ...
    });

Then application may connect to the session again using this token (for example, if incoming call push notification is received):

    var connectionOptions = {
        urlServer: url,
        keepAlive: true
    };

    if (authToken) {
        connectionOptions.authToken = authToken;
    } else {
        connectionOptions.sipOptions = sipOptions;
    }

    Flashphoner.createSession(connectionOptions).on(SESSION_STATUS.ESTABLISHED, function(session, connection){
        ...
    });

Known issues

1. It's impossible to make a SIP call if 'SIP Login' and 'SIP Authentification name' fields are incorrect

Symptoms: SIP call stucks in PENDING state.

Solution: according to the standard, 'SIP Login' and 'SIP Authentification name' should not contain any of unescaped spaces and special symbols and should not be enclosed in angle brackets '<>'.

For example, this is not allowed by the standard

sipLogin='Ralf C12441@host.com'
sipAuthenticationName='Ralf C'
sipPassword='demo'
sipVisibleName='null'

and this is allowed

sipLogin='Ralf_C12441'
sipAuthenticationName='Ralf_C'
sipPassword='demo'
sipVisibleName='Ralf C'

2. There are some problems with sound while SIP calls from Edge browser.

Symptoms:

a) The outgoing sound is sometimes abruptly muffled, then it goes normally.

b) The incoming sound is heard only if you speak into the microphone.

Solution:

Switch SILK and G.722 codecs usage off in SIP calls for Edge browser with stripCodecs option:

var outCall = session.createCall({
    callee: $("#callee").val(),
    visibleName: $("#sipLogin").val(),
    localVideoDisplay: localDisplay,
    remoteVideoDisplay: remoteDisplay,
    constraints: constraints,
    receiveAudio: true,
    receiveVideo: false,
    stripCodecs: "silk,g722"
    ...
});

outCall.call();

or with server setting

codecs_exclude_sip=g722,mpeg4-generic,flv,mpv

3. Microphone swithing does not work in Safari browser.

Symptoms: microphone does not switch using switchMic() WCS WebSDK method.

Solution: use another browser, because Safari always uses sound input microphone, that is chosen in system sound menu (hold down the option (alt) button and click on the sound icon in the menu bar). When microphone is chosen in sound menu, Mac reboot is required.

If Logitech USB camers microphone does not work (when it is chosen in sound menu), format / sample rate changing in Audio MIDI Setup and rebooting can help.

4. Outgoing video SIP call cannot be established if INVITE SDP size exceeds MTU

Symptoms: SIP server return 408 Reques timeout when trying to establish video SIP call, audio calls can be established successfully through the same server.

Solution: reduce the number of codecs in the INVITE SDP so that the SDP fit into the packet size defined by MTU (usually 1500 bytes) using the following settings

codecs_exclude_sip=g722,mpeg4-generic,flv,mpv

3. Microphone swithing does not work in Safari browser.

Symptoms: microphone does not switch using switchMic() WCS WebSDK method.

Solution: use another browser, because Safari always uses sound input microphone, that is chosen in system sound menu (hold down the option (alt) button and click on the sound icon in the menu bar). When microphone is chosen in sound menu, Mac reboot is required.

If Logitech USB camers microphone does not work (when it is chosen in sound menu), format / sample rate changing in Audio MIDI Setup and rebooting can help.

4. Outgoing video SIP call cannot be established if INVITE SDP size exceeds MTU

Symptoms: SIP server return 408 Reques timeout when trying to establish video SIP call, audio calls can be established successfully through the same server.

...

_sip=mpeg4-generic,flv,mpv,opus,ulaw,h264,g722,g729
allow_outside_codecs=false

Only codecs supported by both sides of the call should be left, in this case it is VP8 and PCMA (alaw).

5. There is no sound in browser if caller makes an audio+video call and callee responds with audio only

Symptoms: there is no sound in caller browser if caller makes an audio+video call (for instance, using Phone Video example) and callee responds with audio only (for instance, a call to IVR)

Solution: update WCS to build 5.2.1672 and enable video frames generator

generate_av_for_ua=all

6. IVR greeting plays not from beginning if caller makes audio+video call

Symptoms: there is a gap before IVR greeting starts playing if caller makes an audio+video call (for instance, using Phone Video example)

Solution: update WCS to build 5.2.1755 and reduce video frames generator start timeout

generate_av_for_ua=all
rtp_generator_start_timeout=100

7. An excessive video transcoding to VP8 starts when making a call between browsers

Symptoms: a video traffic is receiving from SIP server in H264 codec, but video is playing as VP8 in browser

Solution:

a) add the following parameter to WCS settings

profiles=42e01f,640028

b) add the following parameter if above does not help

codecsproxy_exclude_sip=mpeg4-generic,flv,mpv,opus,ulaw,h264,g722,g729
allow_outside_codecs=false

...

use_h264_packetization_mode_1_only=false