voip

问题 ![give me error : ( ] i tryed every thing ... [2014-11-15 17:56:06 - LoginActivity] C:\Users\Different\workspace\LoginActivity\AndroidManifest.xml:15: error: Error: No resource found that matches the given name (at 'theme' with value '@style/AppTheme'). [2014-11-15 17:56:06 - LoginActivity] [2014-11-15 17:56:07 - LoginActivity] C:\Users\Different\workspace\LoginActivity\AndroidManifest.xml:15: error: Error: No resource found that matches the given name (at 'theme' with value '@style/AppTheme'

问题 Does anybody is aware of implementing VOIP feature in iPhone and iPad. The things for which i need clarity is, By using which third party library/protocol can I implement this feature? Or is there any in built classed available in objective-c which I can make use of? Is there any Apple store accepted iPhone application which uses VOIP implementation? If so where can I get implementation details of it. This is a generic question. Can we use any third party's in our iPhone application or do we

问题 It's difficult to tell what is being asked here. This question is ambiguous, vague, incomplete, overly broad, or rhetorical and cannot be reasonably answered in its current form. For help clarifying this question so that it can be reopened, visit the help center. Closed 7 years ago . i have to write a chat application enable to conference calling with voip, but i dont know anything about this technology at all and i couldnt find anything useful. do you know any useful site which help to begin

问题 Closed . This question needs to be more focused. It is not currently accepting answers. Want to improve this question? Update the question so it focuses on one problem only by editing this post. Closed 3 months ago . I'm looking into implementing a browser-based VOIP solution that uses SIP and WebRTC and that connects to the PTSN. Basically, users give me their SIP credentials and I use WebRTC to acccess their microphone and speakers. On the page, I plan to run an SIP client. How do I

问题 I am trying to find out if it's possible to build a complete IVR application by cobbling together parts from open source projects. Is anyone using a non-commercial VoiceXML implementation to build speech-enabled systems? 回答1: I've tried JVoiceXML in the past and had some luck with it. http://jvoicexml.sourceforge.net/ It's java of course, but that wasn't a problem for my situation. 回答2: Voiceglue (http://www.voiceglue.org/) is an implementation of voicexml using openvxi and asterisk. It may

问题 I have installed SIPDROID . I made three account at sip2sip. After that I have configured application using one of that account. Application works fine. Question is: If I make multiple call to the same phone using SIP is there anyway to handle all calls at a same time? Another incoming call is not displaying on mobile screen while first call is continue. The first call getting picked up but what about the second call which is coming during first call. Even second call is not displaying on the

问题 I have been looking for a way to set up the Android SIP stack to be able to establish a SIP call between two devices on the same network, in an ad-hoc manner. i.e without REGISTERing to a SIP server. I have not been able to get this to work, as the SIP Demo includes server registration, and I cannot get it to make or receive a call without this step. I am not even sure if this is supposed to be possible. The little mention of this I have been able to find is conflicting (some say it can be

问题 I have created an application using https://www.twilio.com/docs/howto/twilio-client-browser-soft-phone as the template. I am trying to log the CallSid for incoming and outgoing calls so that I can tag calls locally and use the Callsid to link between my local data and twilio's call store. I am able to get the CallSid for incoming calls easily with: Twilio.Device.incoming(function (conn) { if (confirm('Accept incoming call from ' + conn.parameters.From + '?')){ connection=conn; conn.accept();

问题 I'm writing a rtp video streamer for android that reads h264 coded data from an Android local socket and packetize it. The thing is that I did it but I keep getting black frames in the client side (Voip). The communication goes like this: Android -> Asterisk -> Jitsi (Osx) (and reverse) There are a few things that I haven't understood yet: 1) Android's mediarecorder gives me a raw h264 stream, How can I know when a NAL starts / ends based on that stream? It doesn't have any 0x000001 pattern

问题 I'm using the SIP stack in Android 2.3.4 and everything is working nicely. However, I need the system to use SSL, and have created the following code: SipProfile.Builder builder = new SipProfile.Builder("user","domain"); builder.setPassword("password"); builder.setOutboundProxy("sip:IPADDRESS:5061;transport=tls"); builder.setProtocol("TCP"); builder.setAutoRegistration(true); me = builder.build(); Intent i = new Intent(); i.setAction(Constants.INCOMING_CALL_INTENT); PendingIntent pi =