voip

问题 Does anyone know how to use Android device's built-in acoustic echo cancellation? It is located somewhere in silicon and is used for GSM/CDMA speakerphone calls. We'd really like to tap into it for a VoIP application instead of rolling our own. Ben 回答1: I was finally able to get echo cancellation to work on my Arm5 (WM8650) processor (Android 2.2). Below are the steps I took. I wrapped Speex with JNI and called echo processing routines before sending PCM frames to encoder. No echo was

问题 Does anyone know how to use Android device's built-in acoustic echo cancellation? It is located somewhere in silicon and is used for GSM/CDMA speakerphone calls. We'd really like to tap into it for a VoIP application instead of rolling our own. Ben 回答1: I was finally able to get echo cancellation to work on my Arm5 (WM8650) processor (Android 2.2). Below are the steps I took. I wrapped Speex with JNI and called echo processing routines before sending PCM frames to encoder. No echo was

问题 I want to make voip applicaton in j2me.I surfed about mjsip2ME. I want to use mjsip with my application. Can any one suggest me for how to do this ? 回答1: There's a tutorial, but it is kinda incomplete at the moment, as it does not contain source code, and some sections are labeled "TO DO". However, it contains the list of methodw, with descriptions and a quick, general help. There's also some info on the web not on the official site, but I hope you're not afraid of chinese :/ 来源： https:/

问题 I want to make voip applicaton in j2me.I surfed about mjsip2ME. I want to use mjsip with my application. Can any one suggest me for how to do this ? 回答1: There's a tutorial, but it is kinda incomplete at the moment, as it does not contain source code, and some sections are labeled "TO DO". However, it contains the list of methodw, with descriptions and a quick, general help. There's also some info on the web not on the official site, but I hope you're not afraid of chinese :/ 来源： https:/

问题 I working on Project a project to make an iOS App to work like Viber/Skype using VoIP Protocol. For this I am using pjsip SDK but I get some Error :( File not found, unknown type name, etc Please help. Thanks in Advance. 回答1: Based upon your screenshot,you have renamed the sample files in pjsipapp.but you have not configured the header paths and libarary paths. In your target click build settings and in search paths add your library paths.Hope it helps 来源： https://stackoverflow.com/questions

问题 I have the following gstreamer Command-line : gst-launch alsasrc ! mulawenc ! rtppcmupay ! udpsink host= 127.0.0.1 port=5555 It records Mono Voice and i can hear it ,if i listen on 5555 port ( echo IP was used). But i need to transmit Stereo . I have also tried my Microphone for stereo-recording Capability using the following command: arecord -vv -fdat voiceFile.wav and it works. Does anyone know how to specify stereo in the gstreamer command? 回答1: The problem is that rtppcmupay does not

问题 I am trying to remove callkit screen once my voip call is disconnected by source or destination. I used this code CXEndCallAction *endaction = [[CXEndCallAction alloc] initWithCallUUID:[NSUUID UUID]]; CXCallController *callController = [[CXCallController alloc] initWithQueue:dispatch_get_main_queue()]; requestTransaction:[CXTransaction transactionWithActions:nil completion:completion]]; But it is not working to close the callkit. Can any one help me to solve this issue? 回答1: You can use this

问题 I'm analyzing VoIP calls on my network For now i'm using a generated .pcap file, but later i'll be listening for this at real time. I'm using tshark, and i can filter some important data pretty easily from the .pcap (like "source ip address and port", "destination ip addr and Port", payload pckt lost, Max Delta(ms),Max Jitter(ms),Mean Jitter(ms)) with tshark -r myfile -q -z rtp,streams What i want to know is: how can i get the sip addrs of a call? (client and server) I can retrieve some sip

问题 CallKit's call directory extension can be used to block standard phone calls. However how is blocking a VoIP call supposed to work with iOS 13? When there is a VoIP call your app receives a VoIP push, but now with iOS 13 Apple are mandating that when the push is received the app must call CXProvider.reportNewIncomingCall() . However when reportNewIncomingCall() is called, the OS displays an incoming call screen (which is not the same as for a regular incoming call). It is apparently not

问题 I am trying to develop an VOIP application for blackberry,after a long surf i came to know about mjsip api.But i have a doubt that is it possible to use this api with blackberry development to create VOIP application.Please anyone knows the answer help me. 回答1: Thanks, it's a nice project you found there! There is an J2ME version MjSipME, and the only thing I can say now for sure is that it compiles with Blackberry without any errors. UPDATE You right, there are missunderstanding with