Εξέδρα (Exedra)
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Cirrostratus Exedra - testing links from browsers to the phone system.


Call, Answer fully or unmute microphoneEnd or reject callAnswer with muted microphone (or mute)type in numbers or keysChange settingsVideo switch
123Start chat
456Join conference call
789start administrator link (to be able to break into calls)
*0#Acall manager control

Click on the little keyboard (on the left) to type in a number and then click the green handset (on the left) to call that number (type the number with an international code eg 441212561306 to phone 0121 256 1306 in the UK)

SIP Webrtc using the Public Switched Telephone Network

This is a test interface to the PSTN. You are welcome to use this to phone any geographic number in the UK (ie not mobiles). These are numbers that start with 01 02 or 03, or 441, 442 and 443 in E164 format.

This service has been kindly established with the partnership of Soho66. Very soon you will be able to use this service to have your own web interface to the phone system. The phone number that is used by this server is a London number of 020 7183 6588. You can phone the current user of this web page by dialing that number. That, of course, may not be you as someone else could have logged in since you did.

You have logged in as user name 'WEBRTCTEST'.

This is an early test to identify how well this works through various firewalls etc. If you wish to dial a number other than the set number in the database above then click on the keyboard and type in the number that you wish to dial. The system is designed to operate primarily on dialing numbers in a Customer database. However, any UK geographic number should work.

When you dial a number the computer should make some noises. The tone changes every time a SIP message is received. It is also possible that the SIP messages will appear on your screen. These include server messages that never arrive otherwise at your browser. When you click on 'make call' (either send tones on the box or the green handset) then the browser contacts a server and a bridge. The server then contacts another server which links to the PSTN.

Note that other people can access the test logs to see how things operate as everyone who uses this page is automatically logged in as the same user.


This has primarily been tested with Chrome. We will make it work with other webrtc enabled browsers, but for now we still have some work to do to make it work properly with Chrome. It has worked on an android mobile phone using Chrome, but there are still some problems.

Video and other SIP Calls.

The browser javascript is based upon SIP/JS and is a full, but modified sip stack. It will, therefore, do video calls via SIP to any other client which can talk ICE and DTLS. Obviously this includes any browser signed onto part of the Cirrostratus exedra network of servers. It has been tested with Linphone. If you are signed on you should not only be able to receive calls from the phone network (if you have unique control on the login which the test system does not do), but also sip calls from the network. This, however, is still a bit flakey at the moment, but has worked from linphone.

Test video call from this page to webrtctest@webrtc.from-the-net.com


Call, Answer fully or unmute microphoneEnd or reject callAnswer with muted microphone (or mute)type in numbers or keysChange settingsVideo switch
123Start chat
456Join conference call
789start administrator link (to be able to break into calls)
*0#Acall manager control

If you click the green button on the communications element above then it will call this web page. Because the RTC connection is from your computer to itself if this doesn't work then the above PSTN connection is unlikely to work.

The Dial Tone

The Dial tone is not the usual mixture of 350hz and 440hz. Instead we have used 293 1/3 hz for the lower note. This makes it a fifth. Then as each sip message comes in the chord goes up a semitone. If the call fails then the chord is placed a semi-tone below where it started. If you don't like this please tell us (when we have set up the comments bit).

Sip Messages

Do not be surprised if you get a message 'unauthorised' or 'authentication required'. This is normal and happens between the two servers. You don't need to do anything for this to work.

More on messages

The messages part includes SIP, local ICE, remote ICE and media messages. These are all the sorts of things that can go wrong when trying to set up a webrtc conversation. If things don't work and you still cannot work out why try using SAWBUCK on your local system to debug the STUN messages. The local ICE is from PeerConnection status changes. The remote ICE reports Stun messages received for this call at the WebRtc-PSTN bridge. We will probably stop some of these message being sent apart from in debug mode, but for the moment it is all in debug mode.

Test system

Please note that this is an alpha test system, not a beta test system. It would not be surprising if it crashes when you try this. Obviously it is changing working day by working day (not so much at weekends) as problems are identified and resolved.

What if it doesn't work

As with many tech things there are a number of potential problems.