voip

问题 I'm interested in building VOIP that actually can be used to call to analog phone using SIP or H.323. But my question is that, is it even possible to build Computer to Phone & Phone to Computer VOIP phone calls with SIP or H.323? Else what is the most common way of achieving this task? I've successfully built an application that i can transfer voices between two computers by using socket, and my guess is that building SIP to communicate with analog phone is quite complicated (even though i

问题 Google introduced Doze and App Standby mode with Android 6.0 OS. Device will enter in Doze mode if device is unplugged and unused for some amount of time and if application stays inactive for some amount of time, that app goes in StandBy mode (Correct me if I am wrong). Can we (developer) know, device entered in Doze mode or application entered in StandBy mode? How? http://developer.android.com/training/monitoring-device-state/doze-standby.html#whitelisting-cases In this link, Google says

问题 I am creating a VoIP application for IOS. i am using the pjsip open source libraries for this . I am able to connect when i configure the pjsip to UDP . But i always get error code 171060 [Error creating transport: Unsupported transport (PJSIP_EUNSUPTRANSPORT) [status=171060]] I get it that somewhere I am making mistake while configuring the file settings for TLS. This is what i am doing. pjsua_transport_config cfg; pjsua_transport_config_default(&cfg); cfg.port = 5061; cfg.tls_setting.ca

问题 I am facing a problem with Audio When using Callkit with WebRTC for VOIP call, While answering the call from Lock Screen. General Functionality : My app activates the audioSession when it's launched. For an incoming call, SDP Offer & Answer are generated and exchanged. Peer Connection is set up. Both audio and video streams are generated, whether it's audio call or video call. Then Call is reported to callkit by using the following code: callProvider.reportNewIncomingCall(with:

问题 I am implementing VOIP application. I use this sample VOIP sample. when a call changes status, I want to send an event from app service to main project in order to update UI, but I can not find any way to do it, I just can send a message from the main project to app service and receive a response. So anyone can show me the way to send message from app service to main project ? 来源： https://stackoverflow.com/questions/39462097/send-events-from-appservice-to-main-project

问题 I created a soft phone that uses the Twilio Javascript tutorial (quickstart), it works ok...my problem is: i have a system that will receive the call from twilio (browser notification) and i want that the user, answer that call on a new window (pop up), that will show only a mute and a hangup button. Thats the problem, the Twilio object were created on the parent window and works ok, but i was unable to "take" the call in the new (child) window, because once a redeclare the twilio object on

问题 I have an Elastix server, which is being used by my desktop calling app and Zoiper App perfectly for calling purposes. However my own app, which is using Android SIP is not working fine and I am unable to locate the real problem. Whenever I call for profile registration, it gets failed and error message of REGISTRATION NOT RUNNING shows in the logs. here is my code snippet: public void InitializeProfile() { if (mSipManager == null) { return; } if (mSipProfile != null){ closeLocalProfile(); }

问题 We have a VOIP application, that records and plays audio. As such, we are using PlayAndRecord (kAudioSessionCategory_PlayAndRecord) audio session category. So far, we have used it successfully with iPhone 4/4s/5 with both iOS 6 and iOS 7 where call audio and tones played clearly and were audible. However, with iPhone 5s, we observed that both the call audio and tones sound robotic/garbled in speaker mode. When using earpiece/bluetooth/headset, sound is clear and audible. iOS Version used with

问题 Im creating an Android App in which the clients can call each other without using a SIP proxy (server). I downloaded the CSipSimple, it has a "local" option in which the clients in a local network can call each other directly. What if I am connected to the mobile network. If i know the public IP of the destination, can I call him direclty? If you can suggest another applications that fulfill the mentioned requirements please do mention them. thanks 回答1: Yes, you should be able to make direct

问题 I have to initiate a voip call through my app using Siri. It is working in demo project but when I am adding the Intents Extension into my existing project then Siri is not working anymore. In system settings the my app is not showing in App Support section. Plist configuration is like: Also see the plist configuration of extension: Whenever i am giving any voice command it's saying "I wish I could, but < app > hasn't set that up with me yet." I have also tried by enabling Siri from