voip

问题 Is it possible to use an Android phone as a simple GSM gateway? The phone would receive a VoiP call using (preferably) Android built-in SIP stack, initiate a GSM call, and bridge audio both ways. After one call is terminated, the other one ends, too. How could I approach the problem? My earlier attempts failed at bridging audio between connections. Is there a SDK supported way of doing this, that I missed? Or do I need to implement some sort of a workaround? 回答1: There are two problems with

问题 I am trying to implement App-to-App calling with Sinch in my IOS app. I have implemented Pushkit in my iOS app with Sinch but the push notification is not working when the app is in background. I have two questions. Do I need another web service to send push notification to my app for incoming app separately or Sinch handles it itself. If it does handle itself then what am I missing in my code. #import "AppDelegate.h" @interface AppDelegate () @end @implementation AppDelegate - (BOOL

问题 Is there any way to make call by just dialing a local IP address? Simply an IP to IP call. How can I do this? What changes should I make in pjsip code? I don't want to register in any server or VOIP provider. The call will happen only in local WiFi, just like SJPhone applications do in Mac (http://www.sjlabs.com/sjp.html). 回答1: You can happily make LAN-only calls with SIP - it is primarily a peer-to-peer protocol, after all. It's a standard part of SIP, in other words. It looks like pjsip

问题 I am looking for the SIP stacks for Android and iPhone. I found plenty of similar questions, which are sometimes quite old... I do not care too much if the solution is commercial (but this is preferred) or open source. So far I found Commercial solution from RADVISION for Android Open source SIPHONE for iPhone LINPHONE which covers both platforms. Gingerbreadhas built-in SIP stack - but seems that it has some limitations and might be removed by MNO... My questions are: Does anybody has good

问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 5 years ago . I need to add the following reference to my C# project, MyPhoneCRMIntegration.dll the directory of this file is C:\ProgramData\3CXPhone for Windows\PhoneApp , how can I manage to get this in to my C# project? The instructions on the site I've got from are: Create a new project. Select the .NET language of your

问题 I want to integrate language converter in VoIP based iOS client app which will translate real time incoming audio stream to other selected languages based on user choice, I am using PjSip open source library to support VoIP Calls. For language translation I want to use speech to text and text to speech open source library. Now I am facing following issues : How to catch the incoming audio stream in PJSip ? How to send the converted audio stream which user can listen like in normal audio call

问题 I want to make a custom sipdroid client by using reverse byte order. I think that makes other Voip clients cannot decode these data. So I read the code of the SipDroid. I found RTP data goes this way: 1. AudioRecord.read(originalPCM) 2. encode(originalPCM, encodedData) 3. rtp_socket.send(rtp_packet) //the encodeData is rtp_packet's data part And the other side is: 1. rtp_receive(rtp_packet) 2. decode(encodeData, PCMData) //the encodeData is rtp_packet's data part 3. AudioTrack.write(PCMData)

问题 I am trying to run test VoIP program as given in http://www.pjsip.org/pjsip/docs/html/page_pjsip_sample_simple_pjsuaua_c.htm I am trying public SIP servers present at http://www1.cs.columbia.edu/sip/servers.html But I always get error message as 14:33:25.515 pjsua_acc.c SIP registration failed, status=408 (Request Timeout) I never used SIP before, I am not able to guess where the problem is. Is there any simple way to test SIP servers? Does anybody know public free SIP server that works?

问题 I am getting following error while building pjsip as mentioned in this link. When I run the make command: In file included from /usr/lib/gcc/x86_64-linux-gnu/5/include/errno.h:28:0, from ../include/pj/compat/socket.h:131, from ../src/pj/ioqueue_select.c:38: ../src/pj/ioqueue_common_abs.c: In function ‘pj_ioqueue_recv’: ../include/pj/compat/os_auto.h:149:31: error: ‘EAGAIN’ undeclared (first use in this function) #define PJ_BLOCKING_ERROR_VAL EAGAIN ^ Edit: OS Name: Ubuntu 16.04 LTS && 64 bit

问题 Closed. This question is off-topic. It is not currently accepting answers. Want to improve this question? Update the question so it's on-topic for Stack Overflow. Closed 7 years ago . I have a new project on hand on which is quite complex as i am not aware of its domain. My requirement is this, We have list of telephone numbers and from this list i need to check which telephone number is active/real. How would you program a telephony or IVR system to automatically dial phone numbers on this