voip

Twilio: cannot rename subdomain null for SIP termination

由 杀马特。学长 韩版系。学妹 提交于 2019-12-10 23:32:30
问题 In relation to SIP registration, simply trying to add termination URI: Termination URI Configure a SIP Domain Name to uniquely identify your Termination SIP URI for this Trunk. This URI will be used by your communications infrastructure to direct SIP traffic towards Twilio. When you point your infrastructure toward this URI, Twilio uses a Geo DNS lookup to intelligently direct your traffic to our closest POP. Learn more about Termination Settings I can add <foo>.pstn.twilio.com fine, but get

Pjsip iOS How to transmit sound to receiver side and record it?

由 十年热恋 提交于 2019-12-10 23:29:10
问题 func startSipRecording(caller: String, callid: pjsua_call_id) -> (started: Bool, startDate: NSDate?) { var status = pj_init() if status != PJ_SUCCESS.rawValue { return (false, nil) } cpFec = pjsua_data().cp /* Must create a pool factory before we can allocate any memory. */ pj_caching_pool_init(&cpFec!, &pj_pool_factory_default_policy, 0) status = pjmedia_endpt_create(&cpFec!.factory, nil, 1, &med_endpt ) if status != PJ_SUCCESS.rawValue { return (false, nil) } pool = pj_pool_create(&cpFec!

Service for sending SMS and making Voice Calls from Web site and Desktop application

由 為{幸葍}努か 提交于 2019-12-10 18:11:45
问题 We plan to integrate sending of SMS and making calls to our desktop and web applications. Both are written in Java. As for only sending SMS we know about great gateway from Clickatell. But ideally, we would like to use one service similar to it, but which supports Voice Calls and SMS. What service/gateway could you recommend? Here are our main requirements: Reliable Work world wild (if not, at least Europe operators must be supported) Providing external API (for SMS) and reusable components

Android SIP - recursive attempt to load library “/system/lib/librtp_jni.so”

由 狂风中的少年 提交于 2019-12-10 17:57:14
问题 I see above log message when initiating SIP call from Android app using Android SIP SDK. Here's the line which causes it: mSipManager.makeAudioCall(mSipProfileLocal, mSipProfilePeer, listener, 20); I don't think there's anything wrong (at least related to the above message) in that line. But anyway, after that method is called, I see recursive attempt to load library "/system/lib/librtp_jni.so" And SIP call never gets established. Also, I don't receive any error messages/exceptions - nothing.

What Request URL for Voice in TwiML App setup should I use when I develop on localhost?

由 大憨熊 提交于 2019-12-10 17:49:14
问题 I am creating an app in ASP.NET-MVC where I can call phone number from browser. To do that I need to create capability token like in the sample: var capability = new TwilioCapability(accountSid, authToken); capability.AllowClientOutgoing(appSid); capability.AllowClientIncoming("jenny"); string token = capability.GenerateToken(); The appSid is an identifier of TwimlApp created on my Twilio account as described here: https://www.twilio.com/help/faq/twilio-client/how-do-i-create-a-twiml-app I

iOS Backgrounding Not Working

由 余生颓废 提交于 2019-12-10 12:32:39
问题 I am in the process of writing a VoIP application for iOS but when App is in background it stops accepting calls. When the app is active again all the queued up messages start getting processed. The following is what I have done. When building the app I add Voice over IP as well as Audio and AirPlay to the plist file. Then I mark the websocket connection with NetworkServiceTypeVoIP as you can see here. I have not set the keep alive timeout handler because registration doesn't matter if the

PJSUA/PJSIP - Unable to increase support to 32 accounts/transports/calls

由 谁说我不能喝 提交于 2019-12-10 11:50:39
问题 I've tried various attempts using config_site.h during build, and had little to no improvement... Still stuck at 8 accounts. Code is: import pjsua lib = pjsua.Lib() lib.init() lib.start() transport = lib.create_transport(pjsua.TransportType.UDP) for x in range(10): lib.create_account_for_transport(transport) And: Assertion failed: (pjsua_var.acc_cnt < (sizeof(pjsua_var.acc)/sizeof(pjsua_var.acc[0]))), function pjsua_acc_add, file ../src/pjsua-lib/pjsua_acc.c, line 401. 回答1: This issue seems

How to show double height green statusbar (In-Call) in foreground app on device?

由 梦想的初衷 提交于 2019-12-10 02:56:32
问题 There's a lot of questions here asking for displaying a red recording bar while in background. It's totally clear I should use AVAudioSession category AVAudioSessionCategoryPlayAndRecord for that. My question is how can I display a green In-Call bar (or at least red bar) in a foreground app when having an active VOIP call in my app? So I could return to call UI tapping a statusbar area, just like Whatsapp or Skype does. What I've already tried: voip and audio modes in UIBackgroundModes key in

用freeswitch架构自己的VOIP网络电话

由 拈花ヽ惹草 提交于 2019-12-09 20:48:08
目标：通过手机拨号的方式，达到手机与手机、手机与电脑之间的语语音或视频通讯，传输数据全部通过互联网。 附加要求：除了实现一对一通话，还必须实现群组呼叫，其中群组呼叫分两种：1、发起呼叫时，所有的群组内的成员都响铃，但某个人接听时，其它人自动挂断；2：发起呼叫时，所有群组的成员都响铃，每个人可以按接听进入群聊； 实现方式：服务端用开源freeswitch架构软交换服务器；客户端目前已经测试通过的有： android开源客户端：sipdroid、mmsdroid； 电脑PC客户端：x-lite(不开源) ,MicroSIP(开源) 苹果手机开源客户端idoubs正在测试中，不过理论没有问题。 目前以上目标和附加要求都已经实现，如果大家想了解细节，可以@我，如果大家想直接使用我的公网freeswitch服务端来测试的话，也可以向我要测试账号和密码。 来源： oschina 链接： https://my.oschina.net/u/1024047/blog/131614

用freeswitch架设局域网内sip电话

由 こ雲淡風輕ζ 提交于 2019-12-09 20:31:19
FreeSWITCH 是一个电话的软交换解决方案，包括一个软电话和软交换机用以提供语音和聊天的产品驱动。FreeSWITCH 可以用作交换机引擎、PBX、多媒体网关以及多媒体服务器等。 首先保证已经有libjpeg-devel,libtoo,libncurses5,libncurses5-dev git clone git://git.freeswitch.org/freeswitch.git cd freeswitch ./bootstrap.sh 如果要测试IVR的话，需要修改modules.conf，找到mod_flite的行，把它的注释去掉 ./configure make make install 然后需要安装语音文件 sounds-install moh-install (8 kHz) hd-sounds-install hd-moh-install (16 kHz) uhd-sounds-install uhd-moh-install (32 kHz) cd-sounds-install cd-moh-install (48 kHz) 选一个，8 kHz是普通电话的音质，越高越好 make sounds-install moh-install 下载安装完后就算完成了 运行/usr/local/freeswitch/bin/freeswitch开启服务器，要关闭的话
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