voip

Force gstreamer appsink buffers to only hold 10ms of data

泄露秘密 提交于 2019-12-11 10:17:10
问题 I have a gstreamer pipeline which drops all of its data into an appsink: command = g_strdup_printf ("autoaudiosrc ! audio/x-raw-int, signed=true, endianness=1234, depth=%d, width=%d, channels=%d, rate=%d !" " appsink name=soundSink max_buffers=2 drop=true ", bitDepthIn, bitDepthIn, channelsIn, sampleRateIn); Which usually looks something like, autoaudiosrc ! audio/x-raw-int, signed=true, endianness=1234, depth=16, width=16, channels=1, rate=16000 ! appsink name=soundSink max_buffers=2 drop

Auto dial out issue in asterisk

故事扮演 提交于 2019-12-11 08:53:35
问题 I am applying an auto dial in asterisk using .call file My a.call Channel: DAHDI/g0/09********* MaxRetries: 1 RetryTime: 600 WaitTime: 30 Context: outgoing Extension: 10 Priority: 1 My problem is that every time above number is called by same number means even if i change the dialled number(receiver number above) the caller number is same. How can i set the caller number in an outgoing call? Thanks in advance. 回答1: You can use Channel: DAHDI/g0/09********* MaxRetries: 1 RetryTime: 600

Detecting codec used in RTP stream (for dynamic PTs)

余生长醉 提交于 2019-12-11 07:29:14
问题 Is it possible to detect the codec used in an RTP stream by analyzing the RTP stream alone? I know about the payload type (PT) field in the RTP header -- that can be used to identify codecs that have statically assigned PT numbers. What about the other codecs that use the dynamic PT number range? Are there any heuristics to detect the type of the payload? I'm interested in all kinds of solutions, even if the detection works for one codec only. 回答1: Sure, you can sniff around the packet to see

Integrate pjsip in iOS app

空扰寡人 提交于 2019-12-11 07:22:26
问题 I am Goosip/AKSSIP for pjsip wrapper but not able to connect to the server. Can any one guide me to implement pjsip in iOS project. Thanks! 回答1: To Integrate PJSIP on iOS: step 1: First Download PJSIP Project from their website. http://pjsip.org/download.htm step 2: Extract the files in your computer. step 3: Go to the folder pjsip2.6/pjlib/include/pj/ step 4: create config_site.h file step 5: add the following lines into the config_site.h file and save the file #define PJ_CONFIG_IPHONE 1

How to fix leak of SocketStream with service type set to VoIP?

倖福魔咒の 提交于 2019-12-11 07:19:58
问题 Xcode reporting SocketStream object is leaking after streams closing. Leak is reliably reproducible on iPhone, but not on Simulator or Mac. Streams are created with CFStreamCreatePairWithSocketToCFHost , then used, closed and released, leaving another SocketStream every time. I checked retain history in Instruments, and found out that every time read and write streams are closed SocketStream has retain count of 2. (my guess is these two retains are done in

SINCH :: Session Management for App to App Call

ぃ、小莉子 提交于 2019-12-11 06:26:28
问题 I am developing an app with SINCH Platform for VOIP feature. I am performing an app to app call . I am aware that if Callback URL of application Server is set on SINCH Dashboard then it will not trigger DICE (Disconnect Call Event callback) [refer here] event for App to App Call. But I have to manage all call details with my App Server. How Can I manage it from my Mobile App? (Like how can i notify my app server that last call was about of 50 seconds? So my server can manage relevant credit

CallKit with two incoming calls

前提是你 提交于 2019-12-11 06:13:20
问题 I develop a VoIP app with CallKit. In the case that the user receives two incoming calls and he/she accepts one of them the other one will be closed. For example report both incoming calls: reportInComingCall uuid: 70D506FB-6A9D-4111-8828-35DB8F330A26 reportInComingCall uuid: 129A6D67-AC6A-480E-BCD7-ED14F7961CE5 When the user accepts one of them I get this actions from CallKit: perform action: CXEndCallAction uuid: 129A6D67-AC6A-480E-BCD7-ED14F7961CE5 perform action: CXAnswerCallAction uuid:

Showing in-call status bar (double height) for a custom VoIP app in iOS

可紊 提交于 2019-12-11 06:13:01
问题 I have created a VoIP app in iOS using the Sinch library. I am able to make calls successfully, but am having trouble creating the double-height in-call status bar. By this I mean, my call window appears, and on that window I have a button to leave the view and go somewhere else (WITHOUT hanging up the call) and I would like a green double height in-call status bar to be displayed so that I may tap it and return me to call view. When I send the app to the background, I see the double-height

How to close call kit when call doesn't connect but call kit ui is running in the background

爱⌒轻易说出口 提交于 2019-12-11 05:35:37
问题 It happens many times that when call is started it doesn't connect and I can't figure out where it is getting stuck. When it happens the callkit ui is already started as I can see that green notification bar. How do I close callkit ui in this case when call hasn't been connected but callKit ui is running in the background? 回答1: If call has ended by the remote side prior to connecting, you need to report the ended call to CXProvider object. provider.reportCall(with: callUUID, endedAt: Date(),

Sirikit StartAudioCall Fail

冷暖自知 提交于 2019-12-11 05:29:02
问题 Test Background: Xcode: Xcode8 beta6 iPhone: iOS10 beta6 Language: object-c I have used startAudioCall sirikit in my App to test my app call. However, the sirikit will fail in some situation: If the person's name is a first name in local contact, when I say "Call firstname myApp", siri can start myApp get this contact name successfully But if the person's name is both first name and last name, when I say "Call firstname(or both first name and last name) myApp" , siri will call this person