voip

SINCH setSupportManagedPush(true) is not working

由 a 夏天 提交于 2019-12-25 15:59:09
问题 I have been using SINCH SDK's VOICE Calling feature for my project and it was working fine before until I update my google play service dependency. I update it from ' com.google.android.gms:play-services-gcm:10.0.1 ' to ' com.google.android.gms:play-services-gcm:10.2.6 ' and it has started giving me error. If I use the older version then it is working fine. But I have to update it to make it compatible with latest firebase dependencies of version 10.2.6 Here is my crash report. 05-25 14:21:44

自主架设VOIP系统(续)

由 旧巷老猫 提交于 2019-12-25 15:54:36
【推荐】2019 Java 开发者跳槽指南.pdf(吐血整理) >>> 接上篇： 四、Nokia E71手机的配置： 功能表-->工具-->设置-->连接-->SIP设置 新建SIP模式 情景模式名称：八戒VOIP 服务情景模式：IETF 默认接入点：belkin54g 公共用户名：8201@114.255.18.132 （修改完成后会自动变为sip:8201@114.255.18.132） 使用压缩：否 注册：始终注册 使用安全机制：否 代理服务器：代理服务器地址：114.255.18.132（修改完成后会自动变成sip:114.255.18.132） 安全域：asterisk 用户名：8201 密码：asterisk 允许宽松路由：是 传输类型：自动 端口：5060 注册服务器：注册服务器：114.255.18.132（修改完成后会自动变成sip:114.255.18.132） 安全域：asterisk 用户名：8201 密码：asterisk 传输类型：自动 端口：5060 配置成功返回后会变成八戒VOIP/已注册 然后再到功能表-->工具-->设置-->连接-->互联网电话 修改默认模式 名称：八戒网络电话 SIP情景模式：八戒VOIP OK，大功告成了。然后大家拨打电话号码，然后别按拨出键，按中间的白方块钮，选呼叫-->互联网电话呼叫，就可以用voip打电话了。

Sending packets from wireshark to audio decoder: Is .raw file array of RTP packets?

由 泪湿孤枕 提交于 2019-12-25 06:05:06
问题 I have captured RTP packets and need to decode the packets/sesssion with G.729.1 Decoder. In wireshark, I filtered the RTP packets, analyzed and saved the session as .raw file. I am using c# streamdecoder for decoding. Its sample provides example how the speech is encoded, saved in buffer and decoded packet by packet. This is the point I am stuck: const Codec usedCodec = Codec.G7291; const int usedSampleRate = 8000; const int usedBitrate = 12200; var dec = new SpeechDecoder(); dec.SetCodec

Generating an outgoing call in asterisk

由 不问归期 提交于 2019-12-25 04:19:09
问题 I am using asterisk 11.9.0 and i want to generate an outgoing call.I found that for outgoing i have to make a .call file and place it in a var/spool/asterisk/outgoing.I am following the link below http://the-asterisk-book.com/1.6/call-file.html#call-file-parameter my code is same as given in the above link,the above example uses only single fixed number to call. My problem is that i have to generate an outgoing to a number fetched from database(outgoing to new number everytime),so how to

Linphone core listener not receiving incoming calls

由 前提是你 提交于 2019-12-25 02:29:56
问题 I was trying to add sip incoming calls with linphone sdk, The registration is successful and I can make out going calls and the call status is logging as expected, but I am not able to receive incoming calls. I am using intent service to handle connection. Here is my code: protected void onHandleIntent(Intent intent) { String sipAddress = intent.getStringExtra("address"); String password = intent.getStringExtra("password"); final LinphoneCoreFactory lcFactory = LinphoneCoreFactory.instance();

AudioUnitInitialize failed with error code 1701737535 'ent?' after alarm interruption

由 旧城冷巷雨未停 提交于 2019-12-24 22:04:44
问题 I am working with VOIP app. The app is working fine with CallKit. I am facing an issue if alarm fires within call. Every time when alarm stop firing (Audio Interruption ends), we are trying to setActive: on AVAudioSession. But it always gives an error with code 1701737535 ie. 'ent?'. The same error occurs when I am trying to initialize Audio Unit. Without using CallKit it's working fine. Anybody faced issue with activating Audio Session when Audio Interruption ends. I am getting different

asterisk silence detection on connected call

由 不想你离开。 提交于 2019-12-24 15:03:40
问题 Sorry in advance if my question makes no sense to you. I am newbie in asterisk, and what I am trying to do is writing a dial plan which can connects 2 soft phone end point (VoIP client end points) and then try to detect silence in ongoing call. I am able to make through call by using following dial plan exten = 100, 1, Answer() same = 100, n, Monitor() same = 100, n, Dial(SIP/client1,15) when I dialed 100, it makes call to client1, which I received gracefully and now call is on going, now I

iOS - Trigger outgoing VOIP Call on clicking the caller in the native iOS Recent call table view

由 |▌冷眼眸甩不掉的悲伤 提交于 2019-12-24 12:05:13
问题 I have implemented the CallKit for my App to trigger/receive both Audio and Video Call within our App by using WebRTC. Which is working seamlessly without any problem. As I see there are 3 ways to trigger a call from Call Kit. Performing an interaction within the app Opening a link with a supported custom URL scheme Initiating a VoIP call using Siri As of now, I don't have any requirement to initiate the call using Siri but we need to initiate the call when the user clicks on any cell in the

How can I mux/demux RTP media from one stream?

由 最后都变了- 提交于 2019-12-24 11:39:09
问题 Currently, I'm finding a lib able to stream video from multiple sources through one RTP Stream (one connection). Anbody have sugesstion on it? Actually, I figured out that Opal 3.8 is VoIP lib, supported RTP/H264. But I don't know whether it can support mux/demux rtp media from one stream? If no, can you give me some suggesstion? Thanks, 回答1: There are a few RTP stacks around and which one you use depends on which language you are going to be developing in, pjmedia is a good cross-platform

Route all the internet traffic from my android voip app to my own vpn server

由 你说的曾经没有我的故事 提交于 2019-12-24 06:45:06
问题 We have set up our own VPN Server and want to route all the traffic from our VOIP android app through this server. But all the solutions I have seen thus far use the vpn service class http://developer.android.com/reference/android/net/VpnService.html, which creates a vpn tunnel for the whole device and not just my application. I want the other apps running on the phone to use the internet as normal while the traffic from our app is routed through our VPN server. Is there anyway to do this? I
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