pjsip

问题 I am using latest PJSIP library in my Android/iOS applications. I want use TLSv1.2 connection in my project. Which TLS version does PJSIP support? From source code it seems to be supporting TLSv1.0. Is there a way to add TLSv1_2 support to PJSIP library? 回答1: I want use TLSv1.2 connection in my project. Which TLS version does PJSIP support? From Configuring PJSIP with TLS: The TLS support in PJSIP requires OpenSSL development kit (headers and libs) to be installed. So PJSIP could offer

问题 I'm using PJSIP/PJSUA2 with Android project. I've built libpjsua2.so shared library for armeabi, armeabi-v7 and x86 with -fPIC flag. The libs are also linked with OpenSSL 1.0.2h statically which were built with -fPIC flag. All libs loads properly on device with Android 4.4 to 6.0 except x86 in which it shows text relocation error. Build script builds two versions of the lib- android-9 (for API level upto 15 and no x86) android-16 (for API level 16+) Why on x86, text relocation is occurring

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

问题 I have setup kamailio 4.2 on an azure instance as server and for client I am using PJSIP library for Android and iOS applications. The voice calls seem to work well when both the devices are connected to the same network, however, either of the device connects to a different network (or when both the devices are in different networks), they are able to register on SIP server, and even call can be triggered and accepted between both the devices but there is no audio heard on either end. * I

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

问题 I am still learning about SIP and all its protocols, specifically trying to integrate PJSIP into an iPhone application to make p2p calls. I have a question about a peer 2 peer connection using PJSUA. I am able to make calls perfectly to other clients on my local network by calling directly using the URI: sip:192. . .*:5060 I am curious if this will work for making direct calls to other SIP URIs that are not on the local network without using server configuration - if not this way, is there

问题 I'm developing a SIP client based on pjsip on Android (i'm using the csipsimple code as a reference for now). WHen I try to register the account I get the following error: Unable to generate suitable Contact header for registration: Unsupported transport (PJSIP_EUNSUPTRANSPORT) My code is almost the same as in here The sipServer string is the ip address of the registrar "192.168...." I have also tried not to specify the transport method like in the following code: int[] accId = new int[1];

问题 I have sucessfully downloaded the source code for pjsip and compiled it for iphone. I can run the iOS example project that comes with the release. The next step for me is to build my own project that uses the pjsip libraries, this i where i start having problems. I have added the same libraries that the sample project uses to my project. I have also set the "Header Search Paths" under "Build Settings" to "${SOURCE_ROOT}" After that i try to include the pjsip library by typning: #include

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 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