Mizu Webphone - Mizu Voip

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hasmute(boolean)Set to false if you don’t need the mute buttonhasredial(boolean)Set to false if you don’t need the redial buttonhasaudio(boolean)Set to false if you don’t need the audio buttonhasincomingcall(boolean)Set to false if you don’t need the popup for the incoming callshaschat(number)0=no1=API only2=SMS3=IM all (default)hasvolume(number)0=no volume controls1=dynamic (default)2=vertical3=horizontal (useful if you disable the multiple lines)Set to false if you don’t need the volume controls buttonvolumeicons(number)0=no1=text (if hasvolume is set to 3)2=icons (if hasvolume is set to 2)displaysipusername(boolean)Set to true to display the “Extension” edit box.Default is false.When sipusername is set (by applet parameter, user input or javascript api) then it will be used as the sip username and the username field will be used only forauthentification. Otherwise the “username” field will be used for both.displaydisplayname(boolean)Set to true to display the “display name” input box.Default is false.
hideusernamepwdinput(boolean)Set to true if you wish to hide the username/password input controls. Default value is false.hasgui(boolean)Set to false if you are not using the java user interface. (When a custom html/js/css or other skin is used). This is an optional setting.Default is true.Other ParametersThese parameters are more rarely used or should be used only if you have at least a minimal technical knowledge (VoIP and/or JavaScript). You should modify only those parameterswhich you fully understand otherwise better if you leave all with the default values (the default values are already optimized for production).codebase(string)This optional attribute specifies the base URL of the applet--the directory that contains the applet's code. If this attribute is not specified, then the document'sURL is used (your html page URL).Use it if you deploy the webphone.jar in a different directory on your webserver (not the same directory as your webpage).For the toolkit deployment use “JAVA_CODEBASE” instead of “codebase”.Default is ‘.’ which means the same directory (the html document directory)These parameters are more rarely used or should be used only if you have at least a minimal technical knowledge (VoIP and/or JavaScript)use_rport(number)Check rport in SIP signaling (requested and received from the SIP server by the VIA header)0=don’t ask (rtport request will not be added to the VIA header)1=use only for symmetric NAT (only when it is sure that the public address will be correct)2=always (always request and use the returned value except if already on public ip)3=request even on public IP (meaningless in most cases)9=request with the signaling, but don’t use the returned value (good If you want to keep the local IP and for peer to peer calls)Change to 0 or 2 only if you have NAT issues (depending on your server type and settings)(Usually use_rport and use_stun should have the same value set)Default is 1use_stun(number)Stun request on startup.-1=force private address (if the client has both private and public IP, than the private IP will be sent in the signaling)0=no1=use only for stable symmetric NAT (recommended)2=use only for symmetric NAT3=always4=use even on public IPChange if you have NAT issues (depending on your server type and settings)(Usually use_stun and use_rport should have the same value set)Default is 1udpconnect(number)Specify whether the UDP have to be connected before sending on the socket. (Some server might require udp connect and in this way the webphone can alwaysdetect its local address correctly. However this should not be used whit multiple servers or separate domain and outbound proxy)0=no1=on init
Page 1 and 2: 2014Mizu WebSIPPhoneA java applet i
Page 3 and 4: AboutThe Mizu WebSIPPhone is a SIP
Page 5 and 6: Demo versionWe are providing a demo
Page 7 and 8: DeploymentAlthough the webphone is
Page 9 and 10: Click to call / VOIP enabled forums
Page 11 and 12: Main ParametersThe parameters can b
Page 13: The following applet parameters are
Page 17 and 18: 1: change status to ring2: start me
Page 19 and 20: Default value is 2. Set to 3 if you
Page 21 and 22: audiodeviceout(string)Audio device
Page 23 and 24: 2=yes3: yes with extra (this might
Page 25 and 26: (number)G.729 codec setting. 0=neve
Page 27 and 28: 3= yes4=alwaysprack(boolean)Enable
Page 29 and 30: Set to 0 to disable call cut off on
Page 31 and 32: 0=no (default)1=local (in the user
Page 33 and 34: Text logs are sent to the following
Page 35 and 36: Applet parameter securityGeneral co
Page 37 and 38: }public String encrypt(String str)
Page 39 and 40: Will connect to the SIP server. Thi
Page 41 and 42: ‣ -1: unknown‣ 0: no‣ 1: not
Page 43 and 44: string API_GetVersion()Return the p
Page 45 and 46: You will usually have to display th
Page 47 and 48: webphoneobj.API_SetParameter("passw
Page 49 and 50: Version 2.4Version 2.6-improvement:
Page 51 and 52: Version 4.0-fix: codec change on re
Page 53 and 54: Do I need to install java on my web
Page 55 and 56: Select UDP (and not TCP or TLS) for
Page 57 and 58: If the java (JVM or the browser) is
Page 59 and 60: 1. speex wideband (enabled low prio
Page 61 and 62: wp_tel.serveraddress = ''; // yours
Page 63 and 64: theApplet = theApplet.getSubApplet(
Page 65:
9. import Cross Root CA: -keytool -
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