voip

问题 We are developing calling app using Sinch Sdk . I want to remove the uploaded APNS certificate from App created in Sinch Dashboard. I am not finding any way to remove existing uploaded certificate from it. NOTE ::- Now sinch have provided "Remove" options nearby uploaded certification for App. 来源： https://stackoverflow.com/questions/40778933/how-to-remove-uploaded-apns-certificates-from-the-sinch-app-dashboard

问题 We have a VoIP app where we are currently using standard push notifications. We would like to update to using PushKit and VoIP push notifications. I'm a bit unsure how to migrate from our current standard APNS setup to the new. Questions: 1) Will our current APNS production certificate be able to send push messages to new VoIP clients? 2) Will our new VoIP push certificate be able to send push messages to existing, standard APNS apps (tokens)? 回答1: Please do refer pushkit demo https://github

问题 I've been coding a SIP client using the Android SDK's native SIP libs. For some reason I cannot get my account to register with the server. Here are the testing grounds: Linux Mint 17 XFCE running a Kamailio Server(MySQL and TLS enabled). Linux Mint 17 Cinnamon running Android Studio (0.8.6). Asus Google Nexus 7 (2nd gen). All of the above are on 192.168.1.xxx The server and account info has been tested with a working SipClient (ZoIPer). The port 5060 has been tested with a telnet command

问题 I have configured Asterisk 13.13.1 with PJProject 2.5.5 and enable PJSIP as SIP driver (without compiling chan_sip). I have the fully configured system and it's working but I have some problems with incoming calls. I have few numbers connected with my host and when I calling from any public number I noticed this info on asterisk remote console: [Feb 24 14:27:16] NOTICE[5291]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"zzzzz" <sip:zzzzz@192.168.34.1>' failed

问题 I am going to make a PTT project on Android. Could you tell me how deep Android supports Voice and Multimedia API (such as RTP,RTSP,VoIP) for developers? 回答1: MediaPlayer supports playing rtsp://.. URLs. Audio and Video are supported. Check media format support to see which codecs are supported. MediaPlayer internally automatically handles RTSP and RTP, so there is not much you need to handle. OTOH it does not give any low-level control over this process. About VoIP: Android only consumes

问题 Earlier when I was using pjsip 2.7.1 It was working fine. The call recording was perfect. but now I have installed pjsip 2.9. It is crashing on pjmedia_conf_connect_port. SIGABRT because of pjsua_var.mconf. I don't have any idea when it allocated in pjsip. Please explain and help to solve this issue. Thanks in advance I tried to create a media conference before it was using in the recording. but it ended up with no audio. +(void)startRecordingForCalleeId:(NSString *)calleeId andCallId:(int

问题 What are the available API or existing technologies on android that can help one make call using the internet, more like VOIP. I checked out documentation online including this . Kinda confused where to start. Can anyone help with a better guideline ? Thanks. 回答1: PJSIP is one of the better media libraries available on Android. you may want to google the pro and cons of this vs the native android SIP Api and based on your project pick one over the other. There are a few others, but PJSIP is

问题 I'm using FreePBX and have this configuration in extensions_custom.conf so that I can receive a notification via Pushover . [macro-dialout-trunk-predial-hook] exten => s,1,System(/usr/bin/sendpush.php "Call from ${CALLERID(num)} to ${OUTNUM}") I also need to receive notifications on incoming calls, but can't figure it out on what context should I apply it. (If it makes any difference, I'm using 4 trunks and want notifications from all of them) 回答1: Solved by just adding: [ext-did-custom]

问题 With sipster/pjsip sucessfully installed I would like to follow through the basic sipster example and record the sound from a sip connection to that sip server. Now if I run this on a local arch linux machine the sound gets recorded fully, but if I do the same on a docker machine, that I start with docker run -p 5060:5060/udp -it myContainer the SIP connection works, but there is no sound recorded. A friend told me that SIP uses RTP for sound transport and that this protocol binds UDP ports

问题 I have my CMS built in the php, now I want to integrate Voip call on when admin click on the any user's phone number, but I didn't find any solution to get it done. All I have a SIP details of admin and the second user. I did tried the example shown in below link but not working: https://level7systems.co.uk/en/click_to_call_with_php_sip/ it always returning No final response in fr_timer seconds 回答1: After a lot of research I have found that SIPML is the solution of my problem, it gives pre