11.07.2015 Views

Mizu Webphone - Mizu Voip

Mizu Webphone - Mizu Voip

Mizu Webphone - Mizu Voip

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Due to the browser/java stacks, the webphone has around 10 millisecond more delay compared to native application. This is not noticeable by the users.The remote party hear itself back (echo)To solve the aec issues you need to set the “aec” applet parameter to 2. (for better quality also set the “denoise” applet parameter to 1). Make sure that the dll’s(shipped in the mediaench.zip folder) can be downloaded from your server (there are no warnings in the logs regarding aec initialization). You should store themediaench.dll and mediaenchx64.dll near the webphone.jar file on your server and enable the dll mime type (or set the “mediaenchext” applet parameter to“jar” and rename the dlls to jar: mediaench.jar and mediaenchx64.jar). The AEC should eliminate more than 90% of the echo with around 92% success rate. (Thisis if a speaker is used. If the user will use a headset than echo is generated only if the headset is broken)The webphone doesn’t register if I am using the Java Script APICurrent version should not be affected by this issue (only before version 3.6).For the Java Script API to work correctly it is important to let the webphone to send the first message to the server before you begin controlling it via the API.This can be done by using the AI_ServerInit function or one of the following applet parameters: register, call, capabilityrequest, natkeepalive. (If you already havethe username/password information on startup, then you can set the “register” applet parameter to true and let the applet to register automatically. Otherwisethe capabilityrequest or natkeepalive applet parameters should be used. This is a workaround for Java security restrictions which will restrict the applet as itwould be unsigned when it is controlled externally from JavaScript.Chat is not workingMake sure that your softswitch has support for IM and it is enabled. The webphone is using the MESSAGE protocol for this from the SIP SIMPLE protocol suite asdescribed in RFC 3428.Most Asterisk installations might not have support for this by default. You might use Kamailio for this purpose or any other softswitch (most of them has supportfor RFC 3428).The webphone doesn’t receive incoming callsTo be able to receive calls, the webphone must be registered to your server by clicking on the “Connect” button on the user interface (or in case if you don’tdisplay the webphone GUI than you can use the “register” applet parameter with supplied username and password)Once the webphone is registered, the server should be able to send incoming calls to it.The other reason can be if your server doesn’t handle NAT properly.Please try to start the webphone with usestun parameter set to 0 and if still not works then try it with 2.If the calls are still not coming, please send us a log from the webphone (set the applet loglevel parameter to 5) and also from the caller (your server or remoteSIP client)What is the best codec?There is no such thing as the "best codec". All commonly used codec's present in the webphone are well tested and suitable for IP calls.This depends mainly on the circumstances.Usually we recommend G.729 since this provides both good quality and good compression ratio.If G.729 is not available in your license plan, than the other codecs are also fine (GSM, speex, iLBC)Otherwise the G711 codec is the best quality narrowband codec. So if bandwidth is not an issue in your network, than you might prefer PCMU or PCMA (bothhave the same quality)Between webphone users (or other IP to IP calls) you should prefer wideband codec's (this is why you just always leave the speex wideband and ultra widebandwith the highest priority if you have calls between your VoIP users. These will be picked for IP to IP calls and simply omitted for IP to PSTN calls)To calculate the bandwidth needed, you can use this tool. You might also check this blog entry: Codec misunderstandingsWhat is the default codec priority?If you doesn't change the codec priorities with the applet parameters, than the default codec order will be the following (listed in priority order):

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