voip

问题 We are developing an android and iOS app which requires number verification. I am doing it using SINCH Verification SDK. For my app, number verification is being done from Server side using REST API call using SINCH REST API . We are not sure what is the default time period (Session Time Period) for SINCH Pin verification . And how can i change it if i wish to change it? Can anyone help me finding out default SINCH Pin Verificaiton Seesion time? Thanks in advance. 来源： https://stackoverflow

问题 Problems: I am using AsteriskNow which running asterisk 2.0 server in VirtualBox. And i want to connect Asterisk with MySQL databases using ODBC modules. But it fails. When i started with asterisk*CLI> odbc show The command prompt shows that "No such command ODBC SHOW" My Objectives: configure ODBC in asterisk to access MySQL from Asterisk's dialplan directly and dynamically. What i did: I installed my AsteriskNow in VirtualBox. The version of asterisk is 2.0, the CentOS version 5.8 final. I

问题 It is very interesting that I can actually make a mVoIP service from the scratch. I finally could make a call using OpenSIPs source through SIP protocol. I used external STUN server that helps me to find users' private IP address behind NAT firewalls. However, the free external STUN server cannot find user's IP address to make a call when a smartphone is on 3G or 4G network. As a programmer, I won't give up to make a mVoIP service. However, I need a help at this time from you guys. Is there a

问题 I am developing an application based on VOIP(SIP) . I am able to send Invite and Invite-Ack between two emulators but now i would like to send RTPpacket after Invite-ack message in SIP call flow. Is it possible to send like that. If yes, please can anyone help me regarding this. After that i m going to integrate this for both send and receive part of SIP video call. Any help would be appreciated lot!!! Thanks in Advance!!!!! 回答1: i think i will have to look at the RTP implementation of

问题 Someone could tell me why this happen? iOS10, iPhone5s, VOIP Siphon pjsip2.5.5. These issue was not appear every time,it came sometimes when you want to make call or answer call (add at:20161226). Error opening sound device: 14:06:03.137742 pjsua_aud.c ....Set sound device: capture=-1, playback=-2 14:06:03.137775 pjsua_aud.c .....Opening sound device (speaker + mic) PCM@16000/1/20ms 14:06:03.137837 coreaudio_dev. ......Using VoiceProcessingIO audio unit 14:06:03.147404 acquire_call enter,call

问题 I am integrating the voip , thus integrated twilio. I also setup the server code on heroku successfully. I create the token successfully, and getting the callback on handler as well. When I run the application on device, then callback in delegate comes Device: <TCPresenceEvent 0x17de3840 name=jenny, available=YES> didReceivePresenceUpdate After some time, below error occurred: [ERROR TCMetricsPublisher] Failed to push call stats, status code: 403 Delegates callback: I got the callback in

问题 I've got a VoIP video application that works fine on MAC, Windows and iOS using the VP8 codec. When I place a call between any of these platforms and Android both sides of the call have a black and white image with red, green and blue squares. The same happens when Android calls Android. I'm compiling VP8 with: ./libvpx/configure --target=armv7-android-gcc --sdk-path=/Applications/adt/ndk --disable-examples --enable-runtime-cpu-detect --enable-realtime-only --enable-neon My question is why

问题 I would like to get raw audio data from the iPhone mic (in NSData format) to stream over a socket. This is not a situation where I can use twilio/etc, as it is a research project. The socket implementation is done (I can send audio files), but I'm having trouble getting the streaming mic data. Here is my attempt: class ViewController: UIViewController, AVCaptureAudioDataOutputSampleBufferDelegate { override func viewDidLoad() { super.viewDidLoad() // Do any additional setup after loading the

问题 Im running a little app that encodes and decodes an audio array with the speex codec in javascript: https://github.com/dbieber/audiorecorder with a small array filled with a sin waveform for(var i=0;i<16384;i++) data.push(Math.sin(i/10)); this works. But I want to build a VOIP application and have more than one array. So if I split my array up in 2 parts encode>decode>merge, it doesn't sound the same as before. Take a look at this: fiddle: http://jsfiddle.net/exh63zqL/ Both buttons should

问题 I'm making a web application which should be able to monitor calls on my Asterisk server. I can connect to ARI with Javascript WebSocket on URL ws://(host):8088/ari/events?app=dialer and it works. The problem is that I only get events from calls that are made over ARI. Calls made from other clients like Zoiper are not registered. On the other hand, Asterisk has AJAM which uses long polling on http://(host):8088/rawman?action=waitevent and it registers calls from all the clients, (ARI, Zoiper