pjsip

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 When navigating through app app got crashed due to pjsip and it's says you must register external thread with pjsip thread. I've also regsitered pjsip with the main thread like this try { mEndpoint.libRegisterThread(Thread.currentThread().getName()); Log.d(TAG, "Registering with thread "+Thread.currentThread().getName()); } catch (Exception e) { e.printStackTrace(); } This crashed happend so randomly and don't exactly know what causing this and I also not calling any pjsip method when that

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 had successfully registered my sip client in pjsua sample app and is receiving incoming calls over UDP. But whenver I try to make an outgoing call it get disconnected with 403 FORBIDDEN error. And the connection created is TCP! I verified the destination uri. it is in the format as sip:mobile-number@sip-server-ip:port here is my makeCall code : public void makeCall(View view) { //some code here.. MyCall call = new MyCall(account, id); CallOpParam prm = new CallOpParam(true); try { call

问题 I have two application, that uses pjsua. But I can not run both at the same time on a single android device. After running second application it crashes with exception: java.lang.Exception: Title: pjsua_transport_create(type, &tcfg, &tid) 06-26 14:49:56.355 Code: 120098 06-26 14:49:56.355 Description: Address already in use 06-26 14:49:56.355 Location: ../src/pjsua2/endpoint.cpp:1519 06-26 14:49:56.355 14:49:56.361 pjsua_core.c PJSUA state changed: INIT --> STARTING 06-26 14:49:56.355 14:49

问题 I am developing a voip app for I-phone using pjsip as sip stack, and i want to add custom headers am able to add, but only one gets added, am stuck i dont under stand whats wrong, below is my code snippet pj_status_t status = PJ_SUCCESS; pj_str_t pj_uri; pjsua_msg_data msg_data; pjsip_generic_string_hdr subject; pj_str_t hvalue, hname; pj_uri = pj_str((char *)uri); for(NSString *key in [headers allKeys]){ NSLog(@"Call.m key value in call %@,%@",key,[headers objectForKey:key] ); pjsua_msg_data

问题 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

问题 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

问题 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:

问题 I am trying to build an app which is using PJSIP library. I am able to establish a connection successfully with UDP and TCP . But when trying to connect with TLS , I receive the following error: Unsupported Transport error (17160) . 回答1: also you need to add the following parameters to your config_site.h. #define PJ_HAS_SSL_SOCK 1 #define PJSIP_HAS_TLS_TRANSPORT 1 and configure pjsip with --with-openssl=your/opensslDir/path and then check the log after it finishes configuring. if you have