pjsip

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 another way of making p2p calls without server configuration? thanks in advance, You can make calls

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]; accId[0] = 1; pjsua_acc_config cfg = new pjsua_acc_config(); pjsua.acc_config_default(cfg); csipsimple

We are facing an issue regarding callKit Framework by iOS. We have to implement following functionalities in app. One to One call (Working fine) . we can end and accept second call (Working fine) . we can hold and accept calls (max 2 calls). we can switch between calls. Hold/Unhold current call. Issue : The issues we are facing are : We are able to accept second call which have no audio when hold and accept. Switch call button from call kit is disabled. We have done following implementation for handling multiple calls : We are reporting new call by following method. - (void

I've been having issues with compiling pjsip for iOS 4.0. I am using the latest trunk version from SVN and keep getting a portaudio error. When using the piedmontwireless guide: http://www.piemontewireless.net/PJSip155_and_iPhoneSDK312 I get a missing separator error in my build.mak file, which would indicate a whitespace/tabbing error, but for the life of me I cannot find it. According to the pjsip mailing lists, you should be able to compile out of the box for iOS 4.0, but I get this error: ../src/pjmedia-audiodev/errno.c:23:26: error: portaudio.h: No such file or directory ../src/pjmedia

To compile PJSIP library for iPhone device, I am using this code make distclean && make clean ARCH='-arch arm64' ./configure-iphone --enable-opus-codec make dep make This code allows me to install my app for single architecture only. To compile pjsip for all the architectures (armv7, armv7s, arm64, i386, x86_64), Which command or tool I can use Steps to compile for all devices. compile for all devices separately merge with lipo eg. -create /libsrtp-arm64-apple-darwin_ios.a /libsrtp-i386-apple-darwin_ios.a -output output_file_name then use output_file_name.a as a universal library for all

I get this error when I try to establish a new call from pjsip: pjsua_aud.c ..Error retrieving default audio device parameters: Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) [status=420006] Exception: Object: {Account <sip:192.168.0.2:54496>}, operation=make_call(), error=Unable to find default audio device (PJMEDIA_EAUD_NODEFDEV) I installed libasound2-dev : sudo apt-get install libasound2-dev , and recompiled pjsip, but still, it gives me the same result .. Am I missing something here? N.B. I use Ubuntu 11.10 and the sound works fine with no problems, so please don't tell me I

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 ). 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 already supports calling a SIP URI directly (as it should!) so you shouldn't need to adjust anything. Just

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: Anyone know any good examples on how to setup a pjsip client to receive messages. I can send messages from the client using: pjsua_im_send(sip_acc_id, &to, NULL, &msgbody, NULL, NULL); to any number. But I have no idea what to do to receive messages into the already registered sip account. Any info would be greatly appreciated. Note: I can only use pjsip and no other library. Edit: Some new stuff I found: http://trac.pjsip.org/repos/ticket/1070 http://www.pjsip.org/release/0.5.4/PJSIP-Dev-Guide.pdf (however all this document says about

可以将文章内容翻译成中文,广告屏蔽插件可能会导致该功能失效(如失效，请关闭广告屏蔽插件后再试): 问题: I'm trying to build pjsip for android with NDK r13b . Standard build like ./configure-android --with-opus=/home/user/pjsip/pjproject/opus-dev-lib works perfect. But i need several TARGET_ABI and all my tries to do it ends with configure-android error: compiler not found, please check environment settings (TARGET_ABI, etc) Commands as TARGET_ABI=armeabi-v7a ./configure-android --use-ndk-cflags --with-opus=/home/user/pjsip/pjproject/opus-dev-lib TARGET_ABI=armeabi-v7a APP_PLATFORM=android-21 ./configure-android --use-ndk-cflags --with-opus=