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Web Client

The simpliest way to attend a meeting session remotely via Meetecho is using the Lite Web Client we devised just for the scope: all you need to join is your browser.

This Web Client is completely JavaScript (jQuery) based, and as such should work fine with most of the existing browsers. We tested it with an extensive list of different browsers on different platforms, and everything went fine. Should you have any problem, make sure you check the Troubleshooting subsection at the end of this page, or contact us for help.

The functionality currently provided by the Web Client are the following:

  • Integrated VoIP (standard alternatives available as well)
  • Chatroom (including whispers/private messages)
  • Visual Feedback
  • Presentation Sharing
  • File Sharing
  • Whiteboard
  • Shared Text Editor
  • Desktop Sharing
  • Polling
  • CoBrowing

Of course, only a subset of the above mentioned features will be made available for IETF meeting sessions, since the focus will mostly be on the chat, the slides, notetaking and the audio/video feed.

The audio/video stream of a conference can be natively embedded in the Web Client by means of a Java applet: nevertheless, considering the standard nature of Meetecho, several other viable alternatives are provided to Web Client users to access the audio and/or video stream of each session. Specifically, both SIP and RTSP URIs are made available, which can be accessed by making use of third party software (or the browser itself, if enabled). This is especially useful in case you don't want or can't run Java applets in your browser. Besides, a few sessions will also have available a WebRTC/RTCWEB point of entry to the audio/video feeds: this feature obviously requires a compliant browser. Finally, alternatives like HTTP and RTMP streams are provided for an easy fruition of the media streams.


Attending a meeting session


To join the meeting session you're interested in, check the agenda in the homepage. All you need to do is click on the related link, which will look like:

You'll be presented with a simple form: just fill in your full name (e.g., John Doe) and e-mail address (e.g., and you'll be ready to go! Your name will be used as your nickname in the Jabber room.


Chatroom and Participants List

The column on the left includes the conference session chatroom and the participants list.

The chatroom is public: just write something in the text area below it and press enter to send your message to the other participants. The chatroom will also include notifications related to other features (e.g. the current slide number, or a new file being shared).

Below it is the participants list. Every time another participant sends you a private message/whisper, a notification appears next to her/his nickname in the list.


Clicking on a specific participant will allow you to have a private conversation (whispers) with her/him. Clicking on the participant again (or pressing ESC while in the text area) will hide the related window, while keeping the chat history.


If you're not interested in the chatroom, but just in the other features, a little blue arrow below the chat will allow you to hide all the left column and give greater room to the tabs: you'll be able to show the chatroom again at any time by pressing on the new arrow that will appear. This new arrow will also change color whenever new events happen in the hidden chatroom.



The column on the right, instead, has a simple tabbed view of all the other available features, that are:

  • Presentation Sharing (Slides)
  • Text Editor
  • Whiteboard
  • File Sharing (Share File)
  • Media Stream alternatives (Audio/Video)

Just click on a tab to access the related feature: selecting a different tab will just hide the feature, not remove it. Each tab is automatically highlighted whenever something relevant happens to the related feature and the tab is not selected. Clicking on the tab to select it will clear the highlight.

The tab selected by default is "Slides" (Presentation Sharing), since it's the most commonly used.



The "Slides" tab provides the Presentation Sharing functionality. By accessing this tab, you'll be able to either watch an ongoing presentation (if available) or present one yourself. Clicking on the current slide (or the placeholder picture, in case there's no ongoing presentation) will change its zoom level.

To watch an ongoing presentation, just wait for any of the participants to upload one: the Web Client will automatically change slide when the presenter triggers it, so you won't have to do anything. The current slide will also be notified in the chatroom.

To actively present a presentation yourself, press the button on top: this will result in a transparent permission request. As soon as you're allowed to present, there are two ways to start a new presentation:

  1. Choosing a previously uploaded presentation: this will allow you to choose from one of the presentations the system is already aware of for this conference. This is especially useful when some or all of the presentations have been uploaded in advance, and allows for a very quick retrieval of the slides that have to be presented.
  2. Uploading a new presentation: this will allow you to choose a presentation from your files (any MS Office/OpenOffice/PDF document is allowed) and upload it. Once chosen, click on "Submit" to actually upload the slides. This will start the uploading process, followed by the server-side processing of the document. This option usually takes a bit longer than the previous alternative, since a real-time processing of the slides is needed before making them available within the conference.

Whichever option you choose, when the operation completes you will be presented with the first slide of the presentation, and numbered shortcuts on the upper side of the slide panel: just click on any number to trigger the related slide (e.g., clicking on "3" will change the current slide to 3) and every other participant will see that slide as well. Shortcuts (Previous slide, Next slide) are available as well.

You can also have the system automatically send a text message on the jabber chatroom whenever a slide is changed to notify users about the number and title of the current slide: this is especially useful for scribing, since it leverages the scribe from the need of writing these events down himself. To do so, just tick the related checkbox.

If for any reason you want to pass the control of the presentation to another participant in the conference, select the related option on top of the slides: a dropdown menu containing a list of all the participants will be presented. Choose the participant you want to be the new presenter, and then press "Ok". This will present the participant with a popup request: in case she/he accepts to control the slides, the control will be passed to the new presenter and you will become a simple viewer. Of course, this also applies to you in case someone else presented the slides and then wanted to pass the control to you.

In case you want to annotate the current slide, press the "Send slide to whiteboard" button: this will send the current slide to a new tab of the whitebarding tool, thus allowing you and other participants to annotate the slides (see the next section for more details). Sending a slide to the whiteboard is only allowed to the presenter.

To stop the presentation and allow other participants to present another one, just click on "Stop".


Text Editor

A shared text editor is also available: writing is allowed by requesting the editing token. This functionality is especially useful for note taking and for sharing ideas (e.g., a TODO list). Only one participant at a time is allowed to edit the document: the other participants will just be notified about updates.

The current editor needs to manually send the changes whenever the updated document needs to be shared with the other participants: doing so will send the changes transparently to the other participants. The current version of the document can also be exported to either PDF, RTF or HTML at any time.



The Web Client also allows you to draw on a shared whiteboard: just choose the form you want to draw, color, stroke, and so on, and then start drawing on the canvas. Everything you draw will be drawed with the other participants exploiting the functionality.

This feature is particularly useful for collaborative editing, e.g. to share ideas on possible diagrams and so on.

As anticipated in the Slides section, it is also possible to automatically import the current slide from an ongoing presentation in a new whiteboard tab, e.g. to annotate the slide or to foster discussion. This would open a new tab in the whiteboard, and set the current slide as background image: from that moment, it's possible to annotate the slides, e.g. highlighting points, marking points to be modified, adding text comments, shapes, and so on.


Share File

This tab allows you to share a file with the other participants in the conference.

To share a file, just press the button and wait for the permission to upload from the server: as soon as it arrives, you will be allowed to pick a file from your computer. Any file type can be chosen, as long as it is not too large. Once you've chosen the file you want to share, click on submit and the Web Client will take care of the rest.

As soon as the file has been shared, a notification will be sent on the chatroom: this applies to files being shared by other participants than yourself as well. Besides, the file will also be added to the "Share File" tab for a quick reference to all the shared files. Clicking on the shared file will allow you to download it and save it wherever you want.



Audio and Video are supported within the system via several alternatives:


  • Do you only need to listen?
    Then the streaming solution is probably the best and easiest approach! A live RTSP stream is always available, so just grab a media player (e.g. VideoLan Client) and listen to what's happening. Live RTMP streams (one for audio only and one for audio+video) are also provided, which can be opened directly within the browser page if your browser has the Adobe Flash plugin installed. An HTTP streaming link is also provided: this stream is encoded using the recent IETF standardized Opus codec, and can be opened directly in the browser if your browser supports the HTML5 <audio> tag and, as anticipated, the Opus codec. Check the middle part of this section for more details.


  • Do you also need to make questions, present remotely and/or show your video?
    Then you can either exploit the embedded audio/video functionality or access the media stream by means of a SIP or PSTN (landline number) phone. Whichever option better suits you, you'll end up in the conference mix and will be able to participate: all options are described in detail in this section.



Integrated Voip

As anticipated, the Web Client does embed an integrated multimedia functionality: this functionality is made available by means of a Java applet which provides the integrated VoIP features. This means that, just by making use of this applet, it's possible to access the audio and video streams as an active participant by just using the browser.

As anticipated, some sessions will also have a WebRTC/RTCWEB point of entry which provides the exactly same functionality but in a standardized fashion. From an UI point of view, the interfaces are the same as the applet, so in case you're interested in exploiting it just refer to the UI-related notes in this section.


With respect to the Java applet, please notice that this applet-based functionality makes a few assumptions:

  1. That your browser correctly supports Java applets (something that can't always be given for granted);
  2. That your network allows VoIP traffic (specifically, RTP, even if an HTTP tunneling for RTP is provided).

The Web client tries to autodetect your browser's support for Java applets: if everything's fine, a green message should follow the Java version line (as in the screenshot above); an orange message means something may be missing but things may be fine anyway; in case Java applets are not supported at all, the Web client will notify that as well, meaning you'll have to access the audio stream using one of the alternatives.


In order to test whether or not that's the case for you, visit the following page:

which will allow you to start a typical echo test application using the same applet that would be used in the conferencing session. In case something doesn't work as it should, you will probably have to rely on one of the alternatives described in this section to access the multimedia stream. Of course, also please contact us to describe your issue, in order to help us tackle any potential problem that may occur to others as well.


IMPORTANT: If you're going to use Firefox on Mac OS, please be aware that due to some well known issues in the community about this configuration (LiveConnect doesn't work as it should) it is highly discouraged that you try launching the applet, since it may cause the applet to fail or even your browser to hang. So, only proceed if you feel brave enough!! :) As an alternative, you might try using Safari instead which should work fine. Also, make sure Java 1.6 is installed and working, since previous versions are known to give problems with applets as well.


For the above-mentioned reasons, the VoIP applet is not started by default: to enable it, click on the related button in the Audio/Video tab. In case you're only interested in the audio stream, uncheck the "Enable video" checkbox to make sure only audio is negotiated. If you know for sure that RTP is filtered in your network, also tick the "Enable RTP-over-HTTP" checkbox, which will try to encapsulate RTP packets over HTTP.

This will trigger a server-side media negotiation: if successful, the applet will be properly setup and placed in a floating, draggable dialog. A permission request will be presented by the browser in order to launch the applet: this is a standard security policy for signed applets. Accept it in order to take advantage of the VoIP applet. Then, another popup will appear asking you if you want to block the media stream: again, this is a standard Java Applet security question but this time, of course, deny by clicking no, otherwise video won't be available.

The applet may take a while to initialize the first time it is ever started: this is because of the jars that need to be downloaded (slightly more than 1 MB) so just wait until it appears and starts working.

Once loaded, if you enabled video the applet will show you the video mix of the currently active participants in the conference, or a static logo picture if no one is contributing video to the mix.

For what concerns audio, by default the applet will put you in listen-only mode in the conference: you'll only be allowed to listen to the other active participants. In order to actively participate, you can make use of the "I want to talk!" button to transparently issue a floor request (i.e. ask for the permission to talk): the same button can be used to release the floor (i.e. mute yourself). Two sliders are provided in order to allow you to change the volume for both outgoing and/or ingoing audio, if needed.

For what concerns video, by default the applet will put you in watch-only mode in the conference. If for any reason you also want other participants to see you in the video mix, press the "Send video!" button (of course, this assumes you have a webcam and the applet supports it). As for audio, this will transparently send a floor request for video. If accepted, your contribution will be added to the mix. By default your own contribution will not be included in the mix you receive: if you want it to be included anyway, press the small "mirror" icon on the right side of the button.

You can stop the VoIP applet anytime by closing the floating dialog: this will only stop the multimedia features, while all the other functionality will keep on working.


In case you feel uncomfortable with exploiting an applet in your browser, or your browser just doesn't support them, the Web Client is aware of other available media streams as well, which are also notified in the Audio/Video tab. As it can be seen in the screenshot above, besides the Integrated VoIP feature several standard alternatives are provided to access the multimedia stream associated with the conference, namely:

  • A WebRTC/RTCWEB point of entry (not always available);
  • An Icecast-based Opus streaming service (not always available);
  • An RTSP-based audio stream (the audio mix of the conference);
  • An RTMP-based audio stream (the audio mix of the conference) and an RTMP-based video stream (in case you're interested in the video feed as well);
  • A SIP URI to actively participate to the conference;
  • One or more telephone numbers to actively participate the conference (not always available).

All these access points provide different viable alternatives to access the multimedia stream: you can just pick the one that most fits your needs, and open the stream using your favourite third-party software supporting the related protocol.



As anticipated, a few selected session will also have available a WebRTC/RTCWEB point of entry. These standardization efforts aim at providing audio/video real-time communication natively integrated within browsers, and as such are perfect for collaboration scenarios as the ones Meetecho provides for the IETF meeting sessions.

That said, the standardization is still WIP, and as such not many browsers do support it, and none of them currently easily interact with each other. Meetecho has a preliminary WebRTC/RTCWEB support, which is known to work with Chrome Canary and the webrtc4all plugin. Firefox Nightly is unfortunately not supported as of yet.

For sessions that do provide a WebRTC/RTCWEB point of entry, just choose the related icon in the audio/video tab and press the button to launch the call. All the other UI-related features (e.g., mute/unmute) are exactly the same as provided for the Java applet, so refer to the previous session for more info about those.


Multimedia Streaming

If you're just interested in listening to the live audio of the conference, the easiest approach is to just use any RTSP-capable player (e.g. VideoLan) to open the provided RTSP stream.

There are also some stream alternatives integrated in the browser itself: two different Flash-based RTMP streams are provided (audio, audio+video), which will open in a floating popup window when selected, and an HTTP+Opus stream (audio only), which will also open in a floating popup window assuming your browser supports the HTML5 <audio> tag.

An example of a floating window for the audio+video RTMP stream is presented in the following picture.


Phone call (SIP/PSTN)

If instead you're also interested in actively participating to the audio/video discussion, you can use any SIP-capable softphone to access the provided URI to be connected to the audio/video mix. The screenshot below shows a call to the provided URI using Linphone on Linux (audio is G.711, video is H.263).












Since in Meetecho both audio and video are moderated via BFCP, we also enabled a DTMF-based interface to BFCP to allow you to make floor request using BFCP-unaware softphones. The following BFCP-related DTMF strings are supported:

  • * 5 1: request/release the audio floor (needed to speak);
  • * 5 2: request/release the video floor (needed to send your video);
  • * 5 3: enable/disable the mirror mode (your own video in the mix).

Other DTMF strings are available as well:

  • * 4 8: lower the incoming audio volume;
  • * 6 8: increase the incoming audio volume;
  • * 7 8: lower the outgoing audio volume;
  • * 9 8: increase the outgoing audio volume.

Make sure your softphone sends DTMF tones via RTP, and not as sound or SIP INFO (which, in case you didn't know, is EVIL!), otherwise they won't be captured.

Instructions for accessing the media stream using a PSTN phone are exactly the same: to get the audio floor use the same DTMF string as documented above (*51).



Leaving a conference session

To leave an ongoing conference session, just click on the related button above the tabs, and you will be brought back to the conferences list.


Leaving the Web Client

To cleanly leave the Web Client (that is, without just closing the browser ;-) ), a link in the upper-left corner of the page allows you to leave the session and then the system whenever you want to.

Should your session expire for any reason, the system will automatically remove you in case the account inactive for more than a minute: a refresh is handled automatically in the background by the Web Client itself.



The Web Client is completely JavaScript based: so, before attempting to access the Web Client, make sure JavaScript is enabled in your browser.

Specifically, it is based on a well-known JavaScript library called jQuery. This library takes care of as many as possible browser/OS-related incompatibilities JavaScript usually suffers from, but cannot possibly take care of them all. Besides, our Web Client scripts may add other incompatibilities as well. We have tested the Web Client with several different browsers on different OS, and usually everything worked as it should.

Nevertheless, there's still a chance that things may be broken on some other specific systems as well: should this happen to you, 

contact us to describe the issue, and help us improve the application!


  1. Your Web Client sucks!
    Hey that's not a question!
  2. I can't see any login form!
    The Web Client makes a heavy use of JavaScript, did you enable it in your browser? Besides, is your browser listed as supported by jQuery? A few browsers, as Konqueror for instance, don't seem to support the way jQuery animates objects. Try with a different browser as well, if available.
  3. I keep on being redirected to the login form!
    Make sure your browser is not configured to block session cookies for the Web Client website.
  4. I can't hear anything!
    The audio/video feed alternatives are not started by default, did you pick any in the Audio/Video tab? If you chose the Java applet to attend the meeting, did the applet start fine? If so, did you accept the request to allow the applet to be started? Is sound working in other applets? Is RTP filtered in your network? If so, did you try RTP-over-HTTP encapsulation? Did the echo test work? Make also sure you're not the only participant in the audio mix (since the server won't obviously include your own contribution in the mix sent to you), or that your firewall/NAT/ALG doesn't filter RTP traffic.
  5. When I open the applet, everything hangs and I get an error!
    What is your browser/OS configuration? Which version of Java is installed? Does the echo test work fine instead? If you're using Firefox on Mac OS, please be aware that due to some well known issues in the community about this configuration (LiveConnect doesn't work as it should) it is highly discouraged that you try launching the applet, since it may cause the applet to fail or even your browser to hang. As an alternative, you might try using Safari instead which should work fine. Also, make sure Java 1.6 is installed and working, since previous versions are known to give problems with applets as well.
  6. Why can't I send my video? I'm pretty and I want other people to see me!
    Using the applet, due to some JMF-related constraints at the moment video can only be sent on Linux and Windows machines: so, if you're using Mac OS, that's the reason, sorry... if instead you're on Linux/Windows, make sure the webcam is working and recognized by the applet.
  7. I can't hear anything opening the RTSP/RTMP/HTTP streams!
    The passive streams are only enabled when there are active participants in the audio mix: if all the participants are only using the other tools (i.e. not the audio functionality), no audio stream will be available for streaming until someone joins the audio mix as well. If that's not your case, make sure your player/browser supports what the streams require, e.g., that your external multimedia player supports RTSP and the G.711 alaw codec, for RTSP, or that your browser correctly supports Adobe Flash or the HTML5 <audio> tag for the others.
  8. I can't hear anything accessing the SIP URI!
    Has the call been successfully completed? In some cases, the call may be rejected or fail for some other reason. In case the call was successful, check the codecs enabled in your configuration for both audio and video: for what concerns audio, try falling back to the safest codecs (e.g. G.711, GSM); for video, try and force H.263 as a codec, and either QCIF or CIF as resolutions.
  9. I can't hear/see anything accessing the WebRTC/RTCWEB point of entry!
    WebRTC/RTCWEB is a "work in progress" standard, and so not everything might work as expected. The only browser currently known to work is Chrome Canary, even though that browser too might fail for some versions that introduced known bugs or breaking changes. For other browsers, you might also want to make use of the excellent webrtc4all browser plugin, which provides the same set of features to still-not-compliant browsers.
  10. The DTMF strings in SIP/PSTN do nothing!
    How is your softphone sending DTMF digits? Make sure RFC2833 is being exploited (DTMF in RTP), since we don't handle audio signals and SIP INFOs.
  11. Other questions to follow...
    ... and we will answer them!