pjsip

UnsatisfiedLinkError in pjsip library

走远了吗. 提交于 2019-12-31 05:27:31
问题 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

No audio after incoming GSM call disconnected when using CallKit for pjsip iOS app

懵懂的女人 提交于 2019-12-25 17:07:09
问题 When getting incoming GSM calls, I choose hold and accept option. I get the callback from my controller and I put my VoIP call on hold. Normal GSM call audio looks good. Now, when the user disconnects the GSM call using CallKit interface, I get the callback from my setheld action. I unhold my pjsip VoIP call. Once i recieve deactivate from call kit pjsua_aud.c ..Closing iPhone IO device sound playback device and iPhone IO device sound capture device 17:52:26.666 coreaudio_dev.c ..core audio

How to display pjsip incoming video in iOS?

若如初见. 提交于 2019-12-25 16:44:40
问题 I'm working on a PJSIP-Swift project on iOS. I just compiled PJSIP with video settings to allow to use video. I succeeded to send video to another device by using Pjsip. Now I need to display the incoming stream on my iOS application. I put a View in my ViewController in the storyboard in order to display the incoming video in. But my problem is I have no idea how to bind my view with the incoming video from PJSIP. How to do that ? Thanks. Edit : I think I understand better the problem and

PJSIP not running in Windows Phone 8 background process

烈酒焚心 提交于 2019-12-25 01:39:53
问题 I've studied the Windows ChatterBox sample app for Windows Phone 8 and I was able to write a small app to try out PJSIP. However I've run into a problem with the CallInProgressAgent. When creating an outgoing call, I create a VoipPhoneCall object by calling the RequestOutgoingCall method on the VoipCallCoordinator class, thereafter I create the pjsip call by using the pjsua api. It seems pjsip looses all rtp connectivity as soon as I call the RequestOutgoingCall method. My guess is that pjsip

PJSUA2: Contact header uri length limit

二次信任 提交于 2019-12-25 01:39:02
问题 I'm building an android VOIP application with push notifications support, based on PJSUA2. I need to send the push notification (FCM) token to the server (Asterisk in my case) as contact uri parameter, so that I can retrieve it with a script from the server and send notification to wake up client before sending an incoming call request. I put the parameters in the contact uri parameters with acfg.getSipConfig().setContactUriParams(buildParams(contactParameters)); contactParams is a HashMap

Getting “field has incomplete type” and “conflicting types”

[亡魂溺海] 提交于 2019-12-25 00:42:55
问题 I'm trying build pjsip from source with video support by gcc on ubuntu. After i success full run ./configure and make dep, i run make and i have error below: ../src/pjmedia/ffmpeg_util.c:46:18: error: field ‘codec_id’ has incomplete type ../src/pjmedia/ffmpeg_util.c:148:13: error: conflicting types for ‘pjmedia_format_id_to_CodecID’ ../src/pjmedia/ffmpeg_util.h:23:13: note: previous declaration of ‘pjmedia_format_id_to_CodecID’ was here ../src/pjmedia/ffmpeg_util.c: In function ‘pjmedia

PJSUA2 - Recording call audio to wav file

丶灬走出姿态 提交于 2019-12-24 14:09:07
问题 Using PJSUA2 I'm trying to record an incoming call directly to a wav file but can't hear any audio in the wav file. Code below: void SipCall::onCallMediaState(pj::OnCallMediaStateParam& /*prm*/) { pj::CallInfo call_info = getInfo(); pj::AudioMedia* audio_media = 0; for (unsigned int i = 0; i < call_info.media.size(); ++i) { if (call_info.media[i].type == PJMEDIA_TYPE_AUDIO) { audio_media = static_cast<pj::AudioMedia*>(getMedia(i)); break; } } if (audio_media != 0) { try { pj:

pjsip视频颜色通道异常的问题

点点圈 提交于 2019-12-24 07:20:08
红蓝通道颜色交换 问题描述 原因 解决办法 问题描述 pjsip添加了视频支持。也就是编译的时候添加了ffmpeg。 然后我用h263编码,发现一个问题。 A呼叫B。B中的画面(A的摄像头画面)颜色通道被交换了。A中的画面(B的摄像头画面)是正常的。类似于 RGBA 变成了 BGRA。 然后我用h264编码测试,画面是正常的。 因为pjsip使用的ffmpeg进行的编解码,翻看了好久的源码,都没找到颜色通道的设置,最后我放弃了这里。开始进行 视频通话分辨率的设置。最后意外发现了这个问题的原因。 原因 1,ffmpeg只支持特定的几种分辨率(720 480、1280 720……) 如果你使用了不常用的分辨率(eg:126*45),那么就可能出问题。问题的类型包括但不限于:颜色通道异常,解码后分辨率异常、编码不能整成输出、解码乱码…… 2,ffmpeg里面的编码器只支持几种特定的颜色通道类型。 YUV、RGBA、YUYV、RGB…… 如果颜色编码不正确,也是会出现各种问题。 解决办法 没有什么好办法,一切得依着编码器的脾气。 方法: 1,将分辨率转换成合适的分辨率,再送到编码器。 2,转换对应的颜色通道。 来源: CSDN 作者: 蓝黑墨水 链接: https://blog.csdn.net/lanxiaziyi/article/details/103659785

pjsip sip header configuration

人走茶凉 提交于 2019-12-23 14:56:07
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

How to allow inbound calls in pjsip and Asterisk 13?

巧了我就是萌 提交于 2019-12-23 12:33:52
问题 I have configured Asterisk 13.13.1 with PJProject 2.5.5 and enable PJSIP as SIP driver (without compiling chan_sip). I have the fully configured system and it's working but I have some problems with incoming calls. I have few numbers connected with my host and when I calling from any public number I noticed this info on asterisk remote console: [Feb 24 14:27:16] NOTICE[5291]: res_pjsip/pjsip_distributor.c:525 log_failed_request: Request 'INVITE' from '"zzzzz" <sip:zzzzz@192.168.34.1>' failed