pjsip

Pjsip Crashed on RegState callback

匿名 (未验证) 提交于 2019-12-03 02:03:01
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): 由 翻译 强力驱动 问题: In my class that extends SipAccount class I've onRegState() callback which i receive frequently I've notice that after few callback later at any callback like 4th or 5th time it crashes. this is the code from onRegState() callback @Override public void onRegState ( OnRegStateParam prm ) { Log . d ( TAG , "onRegState..." ); PresenceStatus prsenStatus = new PresenceStatus (); if ( prm . getCode (). swigValue ()/ 100 == 2 ) { Log . d ( TAG , "Going online!" ); listener . onLogIn ( true ); prsenStatus . setStatus ( pjsua_buddy_status .

How to use pjsip on android device in order to have G.729 codec functionality

匿名 (未验证) 提交于 2019-12-03 01:53:01
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 want to use G.729 audio codec on my android application. I have did a lot research on this and came to know that pjsip is most promising solution for this. But I have not much idea about all this. can someone provide me complete steps for using pjsip in existing android application and how can i include support of G729 codec via pjsip . Any help will be appreciated. EDIT : Here is my android.mk file. I want to know that,have i did this right? and how to use those C functions in my java code??? include $(CLEAR_VARS) LOCAL_MODULE := pjsua

pjsip sip header configuration

匿名 (未验证) 提交于 2019-12-03 01:22:02
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): 由 翻译 强力驱动 问题: I am using Sip in my ios projects and siphon classes on top of pjsip sdk . I have no problem with basic configuration and therefore I need to add some custom data to my sip header whenever I make a sip call. I have the following header format pjsua_core.c . TX 1123 bytes Request msg INVITE/cseq=31730 (tdta0x92aa400) to UDP xxxxx: 5060: INVITE sip:xxx9@xxxxxx SIP/2.0 Via: SIP/2.0/UDP xxxxx:xxx;rport;branch=z9hG4bKPjt.fUN05fzpwxbm5zJvjoGSA.bnLvoAHl Max-Forwards: 70 From: sip:xxxx@xxxxx;tag=d1Ww0T4iQNqygphKlqLQ.iNcYx-Cdsb2 To: sip

implementing the SIP functionality in IOS5

早过忘川 提交于 2019-12-02 18:37:46
I want to build an iPhone app which has the option to make calls over SIP (VoIP) but at this moment I have no idea how to start. Does anyone have some information about this topic, or maybe a demo project which I can use to implement the SIP functionality? Thanks in advance! You have a long and interesting road ahead of you. : - ). You will need to choose a SIP Library to build your application on top of. There are many open source SIP Stacks, and some of them even have xCode projects for Mac OS and iOS. You might want to look at ReSIPRocate , PJSip , Sofia SIP , or oSIP/exoSIP . You need to

PJSIP TLS not enable for iPhone

南笙酒味 提交于 2019-12-02 10:37:37
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) . 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 done everything successfully you will be able to build it without any error. You can make sure the

How to resolve the linker error in C++ compiler

我只是一个虾纸丫 提交于 2019-12-02 09:32:24
I have to compile PJSIP in CPP compiler. Because I am integrating an API with PJSIP . It is in CPP . So I have to use g++ instead of gcc . But now I didn't integrate any other API. But I am getting linker error in CPP compiler. If it is C compiler, it is working fine. Error: Undefined symbols for architecture arm: "_crypto_alloc", referenced from: srtp_stream_clone(srtp_stream_ctx_t const*, unsigned int, srtp_stream_ctx_t**)in libsrtp-arm-apple-darwin9.a(srtp.o) srtp_stream_alloc(srtp_stream_ctx_t**, srtp_policy_t const*) in libsrtp-arm-apple-darwin9.a(srtp.o) _srtp_create in libsrtp-arm-apple

UnsatisfiedLinkError in pjsip library

与世无争的帅哥 提交于 2019-12-02 08:42:38
I build Pjsip library and use its sample in android studio . question When I run that i got this error.How could I solve it ? thanks in advance. exception jsip.pjsua2.app E/AndroidRuntime: FATAL EXCEPTION: main Process: org.pjsip.pjsua2.app, PID: 4360 java.lang.UnsatisfiedLinkError: No implementation found for void org.pjsip.pjsua2.pjsua2JNI.swig_module_init() (tried Java_org_pjsip_pjsua2_pjsua2JNI_swig_1module_1init and Java_org_pjsip_pjsua2_pjsua2JNI_swig_1module_1init__) at org.pjsip.pjsua2.pjsua2JNI.swig_module_init(Native Method) at org.pjsip.pjsua2.pjsua2JNI.(pjsua2JNI.java:2351) at org

C - how to resolve this error while using pjsip?

荒凉一梦 提交于 2019-12-01 05:53:06
问题 I am getting this error while using pjsip. How to resolve this? i have mic/speaker in the system but its failing to get the device. http://paste.ubuntu.com/6504337/ /* Create audio device parameter to open the device */ static pj_status_t create_aud_param(pjmedia_aud_param *param, pjmedia_aud_dev_index capture_dev, pjmedia_aud_dev_index playback_dev, unsigned clock_rate, unsigned channel_count, unsigned samples_per_frame, unsigned bits_per_sample) { pj_status_t status; /* Normalize device ID

ios pjsip - play a sound during sip call

让人想犯罪 __ 提交于 2019-11-30 07:51:52
问题 When I am on sip call, sometimes I want to send dtmf digits. To do this I created a custom dial pad which when a key is pressed should play a sound of that key, but it is not playing that sound during a sip call (when there is no call, sound is played). These sounds are played with functions from AudioToolbox.h library ( AudioServicesPlaySystemSound(soundID) ). Is there some property that I need to set up in pjsip (pjsua) or in AudioToolbox library to enable a sound be played during a sip

Pjsip Crashed on RegState callback

不羁的心 提交于 2019-11-29 12:29:51
In my class that extends SipAccount class I've onRegState() callback which i receive frequently I've notice that after few callback later at any callback like 4th or 5th time it crashes. this is the code from onRegState() callback @Override public void onRegState(OnRegStateParam prm) { Log.d(TAG, "onRegState..."); PresenceStatus prsenStatus = new PresenceStatus(); if (prm.getCode().swigValue()/100 == 2) { Log.d(TAG, "Going online!"); listener.onLogIn(true); prsenStatus.setStatus(pjsua_buddy_status.PJSUA_BUDDY_STATUS_ONLINE); } else { Log.d(TAG, "Going offline!"); listener.onLogIn(false);