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Linker Errors?

Just tried to install this plugin, but I'm getting tons of Linker errors. I've adjusted the Build Settings, Valid Architectures, etc, but none of that has helped.

I'm running iOS 8.1 on an iPhone 4S. Am I missing something simple?

Which phonertc API is used?

cordova.plugins.phonertc.call(callOptions);

That does not exist in the rtcphone API, how is that? Do I miss something?

Try to receive call on iOS but not sure how

Hi guys, I'm trying to use sipjs-cordova to build an Cordova App to Cordova App Webrtc communication, just tryout the call from App to Browser and it looks great.

But when I tried assign my uri, authUsername and password to my App it seems not working anymore.
I'm not sure if my code is on the right way to do that.

//------New_SIP.UA------//

window.ua = new SIP.UA({
mediaHandlerFactory: PhoneRTCMediaHandler,
traceSip: true,
displayName: 'myDisName',
uri: '[email protected]',
authorizationUser: 'authUserName',
password: '*****'
});

Thanks!

Video in browser Side

I was testing this project!!
First of all, authors, great job!!

I followed this guide: http://sipjs.com/guides/mobile/cordova/

I run the application in IOS without any problem -> GOOD.
I detect the call in the browser, then i answer -> GOOD

In the Iphone Side i can heard the audio and i can see the remote video ( i see remote and local video) -> GOOD.
But in the browser side, i can heard only the sound, the remote video (iphone) is not working, i see only a black square.

I don't know what it's happening,
If i execute the same html code with two browser, the communication it's correct (i think that is not a html mistake)

I attach the IOS OUTPUT and the Firefox Output and simple picture with the case.

Other simple questions:
What is the meaning about that?
No HTML/Javascript video manipulation.
Can't i adjust the size of the video for example?
Can i upload the application to applestore? I think that i need the library built in arm64, but i'm not sure!

I hope your answer, thanks!!

ios_output.txt
firefox.txt

Problem when integrating onsip-cordova

Hi all I have a problem when I try to integrate onsip-cordova on an cordova project. here is the error when I try to build on an iPhone 7:

ld: warning: ignoring file /Users/sophos/Documents/WorkSpace/Test2/platforms/ios/Test2/Plugins/com.onsip.cordova/libwebrtc_video_coding.a, file was built for archive which is not the architecture being linked (arm64): /Users/sophos/Documents/WorkSpace/Test2/platforms/ios/Test2/Plugins/com.onsip.cordova/libwebrtc_video_coding.a
Undefined symbols for architecture arm64:
"OBJC_CLASS$_RTCEAGLVideoView", referenced from:
objc-class-ref in PhoneRTCPlugin.o
"OBJC_CLASS$_RTCICECandidate", referenced from:
objc-class-ref in PhoneRTCDelegate.o
"OBJC_CLASS$_RTCICEServer", referenced from:
objc-class-ref in PhoneRTCPlugin.o
"OBJC_CLASS$_RTCMediaConstraints", referenced from:
objc-class-ref in PhoneRTCDelegate.o
"OBJC_CLASS$_RTCPair", referenced from:
objc-class-ref in PhoneRTCDelegate.o
"OBJC_CLASS$_RTCPeerConnectionFactory", referenced from:
objc-class-ref in PhoneRTCDelegate.o
"OBJC_CLASS$_RTCSessionDescription", referenced from:
objc-class-ref in PhoneRTCDelegate.o
"OBJC_CLASS$_RTCVideoCapturer", referenced from:
objc-class-ref in PhoneRTCDelegate.o
ld: symbol(s) not found for architecture arm64
clang: error: linker command failed with exit code 1 (use -v to see invocation)

a solution please

Called in wrong state: STATE_INPROGRESS. App crashes when the target party receives call

This error is being thrown when the target number receives the call. The app crashes right away.
error=Failed to set remote answer sdp: Called in wrong state: STATE_INPROGRESS

Tried to comment the code where it crashes but the "Accepted" event is never triggered and the call disconnects within a minute.
HELP please 😞
Tried some work arounds but none of them worked perfectly.

Can't show video on ios

I have audio but I didn't see any video(local and remote). How I can show video on view ios
This is my media options:
var options = {
media: {
constraints: {
audio: true,
video: true
} ,
render: {
remote: document.querySelector('video#remote'),
local: document.querySelector('video#local')
}
}
};

build in Xcode 7.1

i don't build app in Xcode. Help me. Lib "libjingle_peerconnection_objc.a" build not use "bitcode"

Should i install phonertc manually after add this plugin ?

cordova com.onsip.sipjs
cordova add com.sipjs.phonertc
cordova add phonertc ?

plugin installing and run for iOS error

cordova plugin add https://github.com/onsip/onsip-cordova.git
Fetching plugin "https://github.com/onsip/onsip-cordova.git" via git clone
Installing "com.onsip.cordova" for ios
Failed to install 'com.onsip.cordova':Error: "/Users/DnV/Projects/SIP/test3/platforms/ios/cordova/build" not located within project!
at Object.module.exports.common.copyFile (/usr/local/lib/node_modules/cordova/node_modules/cordova-lib/src/plugman/platforms/common.js:55:19)
at Object.module.exports.common.asset.install (/usr/local/lib/node_modules/cordova/node_modules/cordova-lib/src/plugman/platforms/common.js:116:20)
at /usr/local/lib/node_modules/cordova/node_modules/cordova-lib/src/plugman/prepare.js:117:26
at Array.forEach (native)
at /usr/local/lib/node_modules/cordova/node_modules/cordova-lib/src/plugman/prepare.js:116:16
at Array.forEach (native)
at Object.handlePrepare (/usr/local/lib/node_modules/cordova/node_modules/cordova-lib/src/plugman/prepare.js:81:24)
at /usr/local/lib/node_modules/cordova/node_modules/cordova-lib/src/plugman/install.js:599:21
at _fulfilled (/usr/local/lib/node_modules/cordova/node_modules/q/q.js:787:54)
at self.promiseDispatch.done (/usr/local/lib/node_modules/cordova/node_modules/q/q.js:816:30)

SIP.js Should not be included as a part of this Repo

It seems that the core SIP.js library should not be included as a part of this repo because it is not kept up to date. At least it should be optional in case people are already including it via Bower or some other package manager (so it stays up to date).

Compile the WebRTC libraries for other architecture than arm7

Hi,
there is a way can bring me the possibility to compile the WebRTC libraries for other architecture than armb7 like i386 and armv7 64bit ?
Can you explain to me how and i will do.

Thanks.

One-way audio with PSTN to iOS 8

Caller hears sound from an iPad Mini Retina running iOS 8 but no sound from caller.

We're using a barebones Cordova application with only the OnSIP plugin. The index.html is that taken from this projects' readme.

Can provide logs but I don't immediately know what would be useful to anyway.

Any help would be appreciated.

Investigate cordova-plugin-webrtc

I learned on discuss-webrtc of a new project which aims to provide iOS with a proper WebRTC API via Cordova. Perhaps it'd be suitable for use here?

@remotium/cordova-plugin-webrtc

Group conversation

Will it be possible in the near future, that the Cordova plugins supports calling more than one person to establish a conference? Or is this just possible by using a server-side conference?

Thanks and best regards,
Chris

Old plugin in registry

It's confusing that there's this new plugin and also this. Can plugins be deleted from the registry, and are you planning to register the new one?

Support for Multiple Sessions

Howdy!

The latest PhoneRTC 2.0 supports sessions and has a much cleaner API. I'm curious if you have explored migrating to enable Group Chat / Multiple Sessions?

alongubkin/phonertc#75

-Lyle

iOS to Web interoperability?

We're getting an error when our iOS app calls a user that is using Chrome. As soon as the Chrome user answers the call, the iOS app crashes and this error shows up in Xcode:

*** Assertion failure in -[PhoneRTCDelegate peerConnection:didSetSessionDescriptionWithError:], ../Plugins/com.onsip.cordova/PhoneRTCDelegate.m:118

Ignored line: c=IN IP4 199.7.175.171
Ignored line: a=rtcp:53775 IN IP4 199.7.175.171
Ignored line: a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
Ignored line: a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
Ignored line: a=sendrecv
Ignored line: a=rtcp-mux
Ignored line: a=rtpmap:103 ISAC/16000
Ignored line: a=rtpmap:111 opus/48000/2
Ignored line: a=rtpmap:0 PCMU/8000
Ignored line: a=rtpmap:8 PCMA/8000
Ignored line: a=rtpmap:106 CN/32000
Ignored line: a=rtpmap:105 CN/16000
Ignored line: a=rtpmap:13 CN/8000
Ignored line: a=rtpmap:126 telephone-event/8000
Ignored line: a=maxptime:60
Ignored line: a=ssrc:2767636768 cname:t+qt1XNiWOZV/vVd
Ignored line: a=ssrc:2767636768 msid:ZfQEHzqJ0uVxUvhUSRTb5HqK6k3Gjl7zzoC5 ab15e6f7-c0e6-4781-b2a5-8803f93b6df1
Ignored line: a=ssrc:2767636768 mslabel:ZfQEHzqJ0uVxUvhUSRTb5HqK6k3Gjl7zzoC5
Ignored line: a=ssrc:2767636768 label:ab15e6f7-c0e6-4781-b2a5-8803f93b6df1
Error(webrtcsdp.cc:354): Failed to parse: "". Reason: Failed to parse audio codecs correctly.

2014-12-22 16:21:00.294 *** Terminating app due to uncaught exception 'NSInternalInconsistencyException', reason: 'Error Domain=RTCSDPError Code=-1 "The operation couldn’t be completed. (RTCSDPError error -1.)" UserInfo=0x146b9510 {error=SessionDescription is NULL.}'

I think the key line is:

Error(webrtcsdp.cc:354): Failed to parse: "". Reason: Failed to parse audio codecs correctly.

It appears that the codecs aren't compatible?

example code typo: "com.onsip.cordova.Sip.js" should be "com.onsip.cordova.Sipjs"

In the readme and in the guide on http://sipjs.com/guides/mobile/cordova/ there is line of code that results in an exception:
var SIP = cordova.require("com.onsip.cordova.Sip.js");
After some hacking, it appears that "com.onsip.cordova.Sip.js" is a typo and should be com.onsip.cordova.Sipjs
If this is the case, it would be cool to see the readme and website edited to reflect this

SIP INVITE not send

Hi, i'm new here.
I'm trying to make a web app to make call between ios or Android Device via an Asterisk.
For compatibility with iOS i read on sipjs guide that i need to use your plugin.
I've installed it on my web project.
When i work with my Ipad the REGISTER on my Asterisk is OK, but if i make a call the INVITE is not send to the Asterisk. I think it's not send by the iPad.
I work with the latest version of iOS.

Anyone can help me ?

Can no longer submit to App Store due to lack of 64-bit support

projects that include sipjs can no longer be submitted to the app store since there is no 64 bit support.

Build Issue

I installed the plugin as instructed:

cordova plugin add https://github.com/onsip/onsip-cordova.git
cp plugins/com.onsip.cordova/build platforms/ios/cordova/

To an existing project. ( Which was originally using just the PhoneRTC Plugin which I had previously removed by: cordova plugin remove com.dooble.phonertc ).

I set the Xcode settings as the instructions in: alongubkin/phonertc/installation.

I'm not 100% sure if that step is required.

The build failed after a few warnings with:

Undefined symbols for architecture armv7:
  "_OBJC_CLASS_$_RTCICECandidate", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCSessionDescription", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCPeerConnectionFactory", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCVideoCapturer", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCPair", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCMediaConstraints", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCEAGLVideoView", referenced from:
      objc-class-ref in PhoneRTCPlugin.o
  "_OBJC_CLASS_$_RTCICEServer", referenced from:
      objc-class-ref in PhoneRTCPlugin.o
ld: symbol(s) not found for architecture armv7
clang: error: linker command failed with exit code 1 (use -v to see invocation)

** BUILD FAILED **

Which suggests there are some missing files.

I'm very rusty with x-code so any help would be appreciated.

Thanks.

plugin add not woking

hi
plugin add command not woking

$ cordova plugin add https://github.com/onsip/onsip-cordova.git

Error: Failed to fetch plugin https://github.com/onsip/onsip-cordova.git via registry.
Probably this is either a connection problem, or plugin spec is incorrect.
Check your connection and plugin name/version/URL.
Error: npm: Command failed with exit code 1 Error output:
npm ERR! cb() never called!

npm ERR! This is an error with npm itself. Please report this error at:
npm ERR! https://github.com/npm/npm/issues

npm ERR! A complete log of this run can be found in:
npm ERR! /Users/mehdi/.npm/_logs/2017-06-30T05_42_23_873Z-debug.log

Undefined symbols for architecture i386 - ionic build ios

I am having problem using sipjs on ionic. I followed all the steps that in the docs appears. I configured .xcodeproj and install this plugin https://github.com/onsip/onsip-cordova.git

Here is the log I received:

Undefined symbols for architecture i386:
  "_OBJC_CLASS_$_RTCEAGLVideoView", referenced from:
      objc-class-ref in PhoneRTCPlugin.o
  "_OBJC_CLASS_$_RTCICECandidate", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCICEServer", referenced from:
      objc-class-ref in PhoneRTCPlugin.o
  "_OBJC_CLASS_$_RTCMediaConstraints", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCPair", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCPeerConnectionFactory", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCSessionDescription", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
  "_OBJC_CLASS_$_RTCVideoCapturer", referenced from:
      objc-class-ref in PhoneRTCDelegate.o
ld: symbol(s) not found for architecture i386
clang: error: linker command failed with exit code 1 (use -v to see invocation)

** BUILD FAILED **


The following build commands failed:
    Ld build/emulator/Business\ Card.app/Business\ Card normal i386
(1 failure)
Error code 65 for command: xcodebuild with args: -xcconfig,/Users/LTroya/LTroya/Cibersys/ionic_tdp/platforms/ios/cordova/build-debug.xcconfig,-project,Business Card.xcodeproj,ARCHS=i386,-target,Business Card,-configuration,Debug,-sdk,iphonesimulator,build,VALID_ARCHS=i386,CONFIGURATION_BUILD_DIR=/Users/LTroya/LTroya/Cibersys/ionic_tdp/platforms/ios/build/emulator,SHARED_PRECOMPS_DIR=/Users/LTroya/LTroya/Cibersys/ionic_tdp/platforms/ios/build/sharedpch
ERROR building one of the platforms: Error: /Users/LTroya/LTroya/Cibersys/ionic_tdp/platforms/ios/cordova/build: Command failed with exit code 2
You may not have the required environment or OS to build this project
Error: /Users/LTroya/LTroya/Cibersys/ionic_tdp/platforms/ios/cordova/build: Command failed with exit code 2
    at ChildProcess.whenDone (/usr/local/lib/node_modules/cordova/node_modules/cordova-lib/src/cordova/superspawn.js:139:23)
    at ChildProcess.emit (events.js:110:17)
    at maybeClose (child_process.js:1015:16)
    at Process.ChildProcess._handle.onexit (child_process.js:1087:5)

I removed the plugin and built it with the same command without problem. That happens when I add this cordova plugin that this log appears.

This info could helps you out:

Cordova CLI: 5.2.0
Gulp version: CLI version 3.9.0
Gulp local:
Ionic Version: 1.0.0
Ionic CLI Version: 1.6.4
Ionic App Lib Version: 0.3.8
ios-deploy version: 1.7.0
ios-sim version: 4.1.1
OS: Mac OS X Yosemite
Node Version: v0.12.7
Xcode version: Xcode 6.4 Build version 6E35b

I will appreciate any hint to help me to solve this problem 👍

Issue with the Get started example

I followed the instructions of the file README.md to deploy the example on iphone and ipad and it doesn’t work.
I try to debug the javascript by adding alert messages and here what I see:
The event deviceready is not called.
I comment the two lines :
First line: document.addEventListener("deviceready", function() {
Last line });

And displaying a message after the line
var SIP = cordova.require("com.onsip.cordova.Sipjs"); but it doesn’t display.

Also there is a problem with var listeningElement = parentElement.querySelector('.listening'); of the file index.js

SIP Call fail on iOS Device but work on browser

Hi everyone,
first thank you for your plugin. It's very cool and useful.
I have a little problem :

I work with a custom SIP server.
I wrote an example code which works on my browser. But the same code on iOS (with ios specifics settings) doesn't work. Few logs have been generated on XCode but the calling failed.

I tried your example code to make a call with the "sip:[email protected]" and it works.

Here the generated logs :

2015-02-09 15:29:54.320 HelloCordova[1097:683021] DiskCookieStorage changing policy from 2 to 0, cookie file: file:///private/var/mobile/Containers/Data/Application/84EFBDF1-42F8-4D25-88E1-31BE04673F4F/Library/Cookies/Cookies.binarycookies
2015-02-09 15:29:54.465 HelloCordova[1097:683021] Apache Cordova native platform version 3.6.3 is starting.
2015-02-09 15:29:54.466 HelloCordova[1097:683021] Multi-tasking -> Device: YES, App: YES
2015-02-09 15:29:54.473 HelloCordova[1097:683021] Unlimited access to network resources
2015-02-09 15:29:54.477 HelloCordova[1097:683021] 

Started backup to iCloud! Please be careful.
Your application might be rejected by Apple if you store too much data.
For more information please read "iOS Data Storage Guidelines" at:
https://developer.apple.com/icloud/documentation/data-storage/

2015-02-09 15:29:54.730 HelloCordova[1097:683021] Resetting plugins due to page load.
2015-02-09 15:29:54.853 HelloCordova[1097:683021] Finished load of: file:///private/var/mobile/Containers/Bundle/Application/B05C6610-A495-418A-858A-8D26EA525FFC/HelloCordova.app/www/index.html
2015-02-09 15:29:54.999 HelloCordova[1097:683021] UA
WebRtc VoiceEngine codecs:
ISAC/16000/1 (103)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
Warning(webrtcvoiceengine.cc:490): Unexpected codec: PCMU/8000/2 (110)
Warning(webrtcvoiceengine.cc:490): Unexpected codec: PCMA/8000/2 (118)
ILBC/8000/1 (102)
G722/16000/1 (9)
Warning(webrtcvoiceengine.cc:490): Unexpected codec: G722/16000/2 (119)
opus/48000/2 (111)
CN/8000/1 (13)
CN/16000/1 (105)
CN/32000/1 (106)
telephone-event/8000/1 (126)
red/8000/1 (127)
WebRtcVideoEngine::WebRtcVideoEngine
webrtc: Thread with name:Trace started 
webrtc: (vie_impl.cc:133): SetTraceFilter: filter: 8206
webrtc: (vie_impl.cc:138): SetTraceCallback: 
WebRtcVoiceEngine::Init
webrtc: Thread with name:ProcessThread started 
webrtc: CheckPlatform
webrtc: current platform is IOS
webrtc: CreatePlatformSpecificObjects
webrtc: output: kPlatformDefaultAudio
webrtc: iPhone Audio APIs will be utilized
webrtc: AttachAudioBuffer
webrtc:   OS info: iOS
webrtc: Thread with name:CaptureWorkerThread started 
webrtc: output: available=0
webrtc: output: available=0
webrtc: TransmitMixer::SetAudioProcessingModule(audioProcessingModule=0x198f71e0)
webrtc: OutputMixer::SetAudioProcessingModule(audioProcessingModule=0x198f71e0)
WebRtc VoiceEngine Version:
VoiceEngine 4.1.0
Build: Aug  6 2014 12:27:42 ?
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, opus_fec: false, }
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 0
Warning(webrtcvoiceengine.cc:867): SetTypingDetectionStatus(0) failed, err=8003
Error(webrtcvideoengine.cc:1469): webrtc: (voe_audio_processing_impl.cc:1001): SetTypingDetectionStatus: not supported
Adjust agc delta is 0
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
Opus FEC is enabled? 0
WebRtc VoiceEngine codecs:
opus/48000/2 (111)
ISAC/16000/1 (103)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
WebRtcVoiceEngine::Init Done!
WebRtcVideoEngine::Init
WebRtcVideoEngine::InitVideoEngine
WebRtc VideoEngine Version:
VideoEngine 3.55.0
Build: Aug  6 2014 12:27:45 ?
webrtc: (vie_base_impl.cc:68): SetVoiceEngine: SetVoiceEngine
VideoEngine Init done
webrtc: (vie_render_impl.cc:64): RegisterVideoRenderModule: 
Applying audio options: AudioOptions {aec: false, agc: false, ns: true, hf: true, swap: false, typing: false, conference: false, agc_delta: 0, experimental_agc: false, experimental_aec: false, experimental_ns: false, aec_dump: false, opus_fec: false, }
High pass filter enabled? 1
Stereo swapping enabled? 0
Typing detection is enabled? 0
Warning(webrtcvoiceengine.cc:867): SetTypingDetectionStatus(0) failed, err=8003
Adjust agc delta is 0
Error(webrtcvideoengine.cc:1469): webrtc: (voe_audio_processing_impl.cc:1001): SetTypingDetectionStatus: not supported
Adjusting AGC level from default -3dB to -3dB
Aec dump is enabled? 0
Experimental aec is 0
Opus FEC is enabled? 0
Allowing SCTP data engine.
Generating identity.
Error(webrtcvideoengine.cc:1469): webrtc: NumberOfCapabilities is not supported on the iOS platform.
Created VideoCapturer for Front Camera
Failed to find best capture format, fall back to the requested format I420 640x480x30
VAdapt input interval changed from 0 to 33333333
Camera 'com.apple.avfoundation.avcapturedevice.built-in_video:1' started with format I420 640x480x30, elapsed time 1 ms
2015-02-09 15:29:56.344 HelloCordova[1097:683042] WARNING: -[<AVCaptureConnection: 0x19a0f720> setVideoMinFrameDuration:] is deprecated.  Please use AVCaptureDevice setActiveVideoMinFrameDuration
2015-02-09 15:29:56.344 HelloCordova[1097:683042] WARNING: -[<AVCaptureConnection: 0x19a0f720> setVideoMaxFrameDuration:] is deprecated.  Please use AVCaptureDevice setActiveVideoMaxFrameDuration
2015-02-09 15:29:56.346 HelloCordova[1097:683021] addLocalStream 1
2015-02-09 15:29:56.347 HelloCordova[1097:683112] PCO onRenegotiationNeeded.
2015-02-09 15:29:56.353 HelloCordova[1097:683021] THREAD WARNING: ['PhoneRTCPlugin'] took '755.441162' ms. Plugin should use a background thread.
Ignored line: c=IN IP4 0.0.0.0
Ignored line: c=IN IP4 0.0.0.0
Created channel for audio
Setting voice channel options: AudioOptions {}
Set voice channel options.  Current options: AudioOptions {}
webrtc: (remote_bitrate_estimator_single_stream.cc:258): RemoteBitrateEstimatorFactory: Instantiating.
webrtc: (vie_base_impl.cc:167): Video channel created: 0
webrtc: (vie_network_impl.cc:74): RegisterSendTransport: channel: 0
webrtc: (vie_network_impl.cc:137): SetMTU: channel: 0 mtu: 1200
webrtc: (vie_rtp_rtcp_impl.cc:285): SetRTCPStatus: channel: 0 mode: 1
webrtc: (vie_rtp_rtcp_impl.cc:521): SetKeyFrameRequestMethod: channel: 0 method: 1
webrtc: (vie_rtp_rtcp_impl.cc:388): SetNACKStatus: channel: 0 on
Warning(webrtcvideoengine.cc:1469): webrtc: (rtp_packet_history.cc:48): Purging packet history in order to re-set status.
NACK enabled for channel 0
webrtc: (vie_base_impl.cc:211): ConnectAudioChannel: ConnectAudioChannel, video channel 0, audio channel 0
webrtc: (vie_rtp_rtcp_impl.cc:558): SetRembStatus: channel: 0 sender: off receiver: off
webrtc: (vie_rtp_rtcp_impl.cc:590): SetReceiveTimestampOffsetStatus: channel: 0enable: off id: 0
webrtc: (vie_rtp_rtcp_impl.cc:627): SetReceiveAbsoluteSendTimeStatus: channel: 0enable: off id: 0
webrtc: (vie_image_process_impl.cc:187): EnableColorEnhancement: video_channel: 0 enable: off
webrtc: (vie_codec_impl.cc:502): RegisterDecoderObserver for channel 0
webrtc: (vie_capture_impl.cc:106): External capture device allocated: 4097
webrtc: (vie_capture_impl.cc:139): Connect capture id 4097 to channel 0
webrtc: (vie_codec_impl.cc:469): RegisterEncoderObserver for channel 0
webrtc: (vie_rtp_rtcp_impl.cc:571): SetSendTimestampOffsetStatus: channel: 0enable: off id: 0
webrtc: (vie_rtp_rtcp_impl.cc:608): SetSendAbsoluteSendTimeStatus: channel: 0enable: off id: 0
webrtc: (vie_rtp_rtcp_impl.cc:658): SetTransmissionSmoothingStatus: channel: 0 enable: on
webrtc: (vie_rtp_rtcp_impl.cc:558): SetRembStatus: channel: 0 sender: off receiver: off
webrtc: (vie_rtp_rtcp_impl.cc:388): SetNACKStatus: channel: 0 on
Warning(webrtcvideoengine.cc:1469): webrtc: (rtp_packet_history.cc:48): Purging packet history in order to re-set status.
NACK enabled for channel 0
webrtc: (vie_base_impl.cc:301): StartReceive: StartReceive 0
Created channel for video
Improved WIFI BWE called.
webrtc: (vie_network_impl.cc:166): SetBandwidthEstimationConfig: channel: 0
webrtc: (remote_bitrate_estimator_single_stream.cc:258): RemoteBitrateEstimatorFactory: Instantiating.
Session:1051969493015781748 Old state:STATE_INIT New state:STATE_SENTINITIATE Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
2015-02-09 15:29:56.384 HelloCordova[1097:683112] PCO onSignalingStateChange: 1
Setting local voice description
Add send ssrc: 4137417215
Setting receive voice codecs:
ISAC/16000/1 (103)
opus/48000/2 (111)
G722/16000/1 (9)
ILBC/8000/1 (102)
PCMU/8000/1 (0)
PCMA/8000/1 (8)
CN/32000/1 (106)
CN/16000/1 (105)
CN/8000/1 (13)
red/8000/1 (127)
telephone-event/8000/1 (126)
Changing voice state, recv=0 send=0
Setting local video description
webrtc: (vie_rtp_rtcp_impl.cc:590): SetReceiveTimestampOffsetStatus: channel: 0enable: on id: 2
webrtc: (vie_rtp_rtcp_impl.cc:627): SetReceiveAbsoluteSendTimeStatus: channel: 0enable: on id: 3
AddSendStream {id:ARDAMSv0;ssrcs:[2875234910,1748033902];ssrc_groups:{semantics:FID;ssrcs:[2875234910,1748033902]};cname:UmG/becHMv7XQ9q1;sync_label:ARDAMS}
webrtc: (vie_rtp_rtcp_impl.cc:114): SetLocalSSRC: channel: 0 ssrc: 2875234910
webrtc: (vie_rtp_rtcp_impl.cc:114): SetLocalSSRC: channel: 0 ssrc: 1748033902
webrtc: (vie_rtp_rtcp_impl.cc:321): SetRTCPCName: channel: 0 rtcp_cname: UmG/becHMv7XQ9q1
webrtc: (vie_rtp_rtcp_impl.cc:114): SetLocalSSRC: channel: 0 ssrc: 2875234910
Add send ssrc: 2875234910
webrtc: (vie_codec_impl.cc:278): SetReceiveCodec for channel 0
webrtc: (vie_codec_impl.cc:279): Codec type 0, payload type d
webrtc: (vie_codec_impl.cc:278): SetReceiveCodec for channel 0
webrtc: (vie_codec_impl.cc:279): Codec type 3, payload type t
webrtc: (vie_codec_impl.cc:278): SetReceiveCodec for channel 0
webrtc: (vie_codec_impl.cc:279): Codec type 4, payload type u
webrtc: (vie_rtp_rtcp_impl.cc:227): SetRtxReceivePayloadType: channel: 0 payload_type: 96
Buffer latency is 0
webrtc: (vie_rtp_rtcp_impl.cc:474): SetSenderBufferingMode: channel: 0 target_delay_ms: 0
Warning(webrtcvideoengine.cc:1469): webrtc: (rtp_packet_history.cc:48): Purging packet history in order to re-set status.
webrtc: (vie_rtp_rtcp_impl.cc:501): SetReceiverBufferingMode: channel: 0 target_delay_ms: 0
Changing video state, recv=0 send=0
Setting voice channel options: AudioOptions {}
Set voice channel options.  Current options: AudioOptions {}
webrtc: (vie_base_impl.cc:78): RegisterCpuOveruseObserver: RegisterCpuOveruseObserver on channel 0
webrtc: (vie_base_impl.cc:78): RegisterCpuOveruseObserver: RegisterCpuOveruseObserver on channel 0
WebRtcOveruseObserver enable: 0
Local and Remote descriptions must be applied to get SSL Role of the session.
2015-02-09 15:29:56.391 HelloCordova[1097:683021] sendMessage 1
2015-02-09 15:29:56.391 HelloCordova[1097:683021] SENDING MESSAGE: {"sdp":"v=0\r\no=- 1051969493015781748 2 IN IP4 127.0.0.1\r\ns=-\r\nt=0 0\r\na=group:BUNDLE audio video\r\na=msid-semantic: WMS ARDAMS\r\nm=audio 1 RTP\/SAVPF 103 111 9 102 0 8 106 105 13 127 126\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:l66L4fecuxy\/++mT\r\na=ice-pwd:kCmblardPuqpudu8dm1lM105\r\na=ice-options:google-ice\r\na=fingerprint:sha-1 5A:AA:38:4C:6A:FC:98:2C:03:AA:3A:96:6D:5B:18:90:F7:EC:51:08\r\na=setup:actpass\r\na=mid:audio\r\na=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level\r\na=extmap:3 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:111 opus\/48000\/2\r\na=fmtp:111 minptime=10\r\na=rtpmap:103 ISAC\/16000\r\na=rtpmap:9 G722\/16000\r\na=rtpmap:102 ILBC\/8000\r\na=rtpmap:0 PCMU\/8000\r\na=rtpmap:8 PCMA\/8000\r\na=rtpmap:106 CN\/32000\r\na=rtpmap:105 CN\/16000\r\na=rtpmap:13 CN\/8000\r\na=rtpmap:127 red\/8000\r\na=rtpmap:126 telephone-event\/8000\r\na=maxptime:60\r\na=ssrc:4137417215 cname:UmG\/becHMv7XQ9q1\r\na=ssrc:4137417215 msid:ARDAMS ARDAMSa0\r\na=ssrc:4137417215 mslabel:ARDAMS\r\na=ssrc:4137417215 label:ARDAMSa0\r\nm=video 1 RTP\/SAVPF 100 116 117 96\r\nc=IN IP4 0.0.0.0\r\na=rtcp:1 IN IP4 0.0.0.0\r\na=ice-ufrag:l66L4fecuxy\/++mT\r\na=ice-pwd:kCmblardPuqpudu8dm1lM105\r\na=ice-options:google-ice\r\na=fingerprint:sha-1 5A:AA:38:4C:6A:FC:98:2C:03:AA:3A:96:6D:5B:18:90:F7:EC:51:08\r\na=setup:actpass\r\na=mid:video\r\na=extmap:2 urn:ietf:params:rtp-hdrext:toffset\r\na=extmap:3 http:\/\/www.webrtc.org\/experiments\/rtp-hdrext\/abs-send-time\r\na=sendrecv\r\na=rtcp-mux\r\na=rtpmap:100 VP8\/90000\r\na=rtcp-fb:100 ccm fir\r\na=rtcp-fb:100 nack\r\na=rtcp-fb:100 nack pli\r\na=rtcp-fb:100 goog-remb\r\na=rtpmap:116 red\/90000\r\na=rtpmap:117 ulpfec\/90000\r\na=rtpmap:96 rtx\/90000\r\na=fmtp:96 apt=100\r\na=ssrc-group:FID 2875234910 1748033902\r\na=ssrc:2875234910 cname:UmG\/becHMv7XQ9q1\r\na=ssrc:2875234910 msid:ARDAMS ARDAMSv0\r\na=ssrc:2875234910 mslabel:ARDAMS\r\na=ssrc:2875234910 label:ARDAMSv0\r\na=ssrc:1748033902 cname:UmG\/becHMv7XQ9q1\r\na=ssrc:1748033902 msid:ARDAMS ARDAMSv0\r\na=ssrc:1748033902 mslabel:ARDAMS\r\na=ssrc:1748033902 label:ARDAMSv0\r\n","type":"offer"}
Transport: audio, allocating candidates
Jingle:Net[pdp_ip0:10.186.38.104/32:Unknown]: Allocation Phase=Udp
Jingle:Port[:1:0::Net[pdp_ip0:10.186.38.104/32:Unknown]]: Port created
AllocationSequence: UDPPort will be handling the STUN candidate generation.
Adding allocated port for audio
Jingle:Port[audio:1:0::Net[pdp_ip0:10.186.38.104/32:Unknown]]: Added port to allocator
2015-02-09 15:29:56.396 HelloCordova[1097:683112] PCO onIceGatheringChange. 1
2015-02-09 15:29:56.400 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.401 HelloCordova[1097:683112] SENDING MESSAGE: {"state":"GATHERING","type":"IceGatheringChange"}
Transport: audio, allocating candidates
2015-02-09 15:29:56.402 HelloCordova[1097:683112] PCO onIceGatheringChange. 1
2015-02-09 15:29:56.403 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.404 HelloCordova[1097:683112] SENDING MESSAGE: {"state":"GATHERING","type":"IceGatheringChange"}
Transport: video, allocating candidates
2015-02-09 15:29:56.406 HelloCordova[1097:683112] PCO onIceGatheringChange. 1
2015-02-09 15:29:56.406 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.406 HelloCordova[1097:683112] SENDING MESSAGE: {"state":"GATHERING","type":"IceGatheringChange"}
Transport: video, allocating candidates
2015-02-09 15:29:56.407 HelloCordova[1097:683112] PCO onIceGatheringChange. 1
2015-02-09 15:29:56.408 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.408 HelloCordova[1097:683112] SENDING MESSAGE: {"state":"GATHERING","type":"IceGatheringChange"}
2015-02-09 15:29:56.409 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[audio] Index[0] Sdp[candidate:1616667645 1 udp 2122260223 10.186.38.104 60113 typ host generation 0]
2015-02-09 15:29:56.410 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.410 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.411 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"audio","candidate":"candidate:1616667645 1 udp 2122260223 10.186.38.104 60113 typ host generation 0","label":0}
2015-02-09 15:29:56.412 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[audio] Index[0] Sdp[candidate:1616667645 2 udp 2122260223 10.186.38.104 60113 typ host generation 0]
2015-02-09 15:29:56.412 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.413 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.413 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"audio","candidate":"candidate:1616667645 2 udp 2122260223 10.186.38.104 60113 typ host generation 0","label":0}
2015-02-09 15:29:56.414 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[video] Index[1] Sdp[candidate:1616667645 1 udp 2122260223 10.186.38.104 60113 typ host generation 0]
2015-02-09 15:29:56.415 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.415 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.416 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"video","candidate":"candidate:1616667645 1 udp 2122260223 10.186.38.104 60113 typ host generation 0","label":1}
2015-02-09 15:29:56.417 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[video] Index[1] Sdp[candidate:1616667645 2 udp 2122260223 10.186.38.104 60113 typ host generation 0]
2015-02-09 15:29:56.417 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.418 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.418 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"video","candidate":"candidate:1616667645 2 udp 2122260223 10.186.38.104 60113 typ host generation 0","label":1}
Jingle:Net[pdp_ip0:10.186.38.104/32:Unknown]: Allocation Phase=Relay
Jingle:Port[:1:0:relay:Net[pdp_ip0:10.186.38.104/32:Unknown]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0:relay:Net[pdp_ip0:10.186.38.104/32:Unknown]]: Added port to allocator
Warning(turnport.cc:488): Jingle:Port[audio:1:0:relay:Net[pdp_ip0:10.186.38.104/32:Unknown]]: TURN host lookup received error 8
2015-02-09 15:29:56.452 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[audio] Index[0] Sdp[candidate:3776210249 1 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0]
2015-02-09 15:29:56.453 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.453 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.453 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"audio","candidate":"candidate:3776210249 1 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0","label":0}
2015-02-09 15:29:56.454 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[audio] Index[0] Sdp[candidate:3776210249 2 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0]
2015-02-09 15:29:56.455 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.455 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.456 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"audio","candidate":"candidate:3776210249 2 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0","label":0}
2015-02-09 15:29:56.457 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[video] Index[1] Sdp[candidate:3776210249 1 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0]
2015-02-09 15:29:56.457 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.458 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.458 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"video","candidate":"candidate:3776210249 1 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0","label":1}
2015-02-09 15:29:56.459 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[video] Index[1] Sdp[candidate:3776210249 2 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0]
2015-02-09 15:29:56.460 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.460 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.461 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"video","candidate":"candidate:3776210249 2 udp 1686052607 80.12.39.104 60113 typ srflx raddr 10.186.38.104 rport 60113 generation 0","label":1}
Jingle:Net[pdp_ip0:10.186.38.104/32:Unknown]: Allocation Phase=Tcp
Jingle:Port[:1:0:local:Net[pdp_ip0:10.186.38.104/32:Unknown]]: Port created
Adding allocated port for audio
Jingle:Port[audio:1:0:local:Net[pdp_ip0:10.186.38.104/32:Unknown]]: Added port to allocator
2015-02-09 15:29:56.504 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[audio] Index[0] Sdp[candidate:786407181 1 tcp 1518280447 10.186.38.104 56344 typ host generation 0]
2015-02-09 15:29:56.504 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.505 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.505 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"audio","candidate":"candidate:786407181 1 tcp 1518280447 10.186.38.104 56344 typ host generation 0","label":0}
2015-02-09 15:29:56.506 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[audio] Index[0] Sdp[candidate:786407181 2 tcp 1518280447 10.186.38.104 56344 typ host generation 0]
2015-02-09 15:29:56.507 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.508 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.508 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"audio","candidate":"candidate:786407181 2 tcp 1518280447 10.186.38.104 56344 typ host generation 0","label":0}
2015-02-09 15:29:56.510 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[video] Index[1] Sdp[candidate:786407181 1 tcp 1518280447 10.186.38.104 56344 typ host generation 0]
2015-02-09 15:29:56.511 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.511 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.512 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"video","candidate":"candidate:786407181 1 tcp 1518280447 10.186.38.104 56344 typ host generation 0","label":1}
2015-02-09 15:29:56.512 HelloCordova[1097:683112] PCO onICECandidate.
  Mid[video] Index[1] Sdp[candidate:786407181 2 tcp 1518280447 10.186.38.104 56344 typ host generation 0]
2015-02-09 15:29:56.513 HelloCordova[1097:683112] gotICECandidate -- sending message
2015-02-09 15:29:56.513 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.514 HelloCordova[1097:683112] SENDING MESSAGE: {"type":"candidate","id":"video","candidate":"candidate:786407181 2 tcp 1518280447 10.186.38.104 56344 typ host generation 0","label":1}
Jingle:Net[pdp_ip0:10.186.38.104/32:Unknown]: Allocation Phase=SslTcp
All candidates gathered for audio:1:0
Transport: audio, component 1 allocation complete
Transport: audio, component 2 allocation complete
Transport: video, component 1 allocation complete
Transport: video, component 2 allocation complete
Transport: audio allocation complete
Transport: video allocation complete
Candidate gathering is complete.
2015-02-09 15:29:56.556 HelloCordova[1097:683112] PCO onIceGatheringChange. 2
2015-02-09 15:29:56.557 HelloCordova[1097:683112] sendMessage 1
2015-02-09 15:29:56.557 HelloCordova[1097:683112] SENDING MESSAGE: {"state":"COMPLETE","type":"IceGatheringChange"}
Captured frame size 480x640. Expected format I420 640x480x30
VAdapt Input Resolution Change: Previous input resolution: 640x480 New input resolution: 480x640 New output resolution: 480x640
2015-02-09 15:29:56.789 HelloCordova[1097:683042] Reset Ui
2015-02-09 15:29:56.790 HelloCordova[1097:683042] sendMessage 1
Destroying NSS identity
Destroying NSS identity
Warning(webrtcvideoengine.cc:2550): GetStats: sender information not ready.
Session:1051969493015781748 Old state:STATE_SENTINITIATE New state:STATE_RECEIVEDTERMINATE Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
2015-02-09 15:29:56.795 HelloCordova[1097:683112] PCO onIceConnectionChange. 6
2015-02-09 15:29:56.796 HelloCordova[1097:683112] PCO onSignalingStateChange: 5
webrtc: (neteq_impl.cc:112): Deleting NetEqImpl object.
Destroyed channel
webrtc: (vie_base_impl.cc:78): RegisterCpuOveruseObserver: RegisterCpuOveruseObserver on channel 0
webrtc: (vie_codec_impl.cc:485): DeregisterEncoderObserver for channel 0
webrtc: (vie_capture_impl.cc:181): DisconnectCaptureDevice 0
webrtc: (vie_capture_impl.cc:125): ReleaseCaptureDevice 4097
webrtc: (vie_render_impl.cc:135): RemoveRenderer: render_id: 0
Warning(webrtcvideoengine.cc:1469): webrtc: Stop: Not running
webrtc: (vie_network_impl.cc:94): DeregisterSendTransport: channel: 0
Removing video stream 0 with VideoEngine channel #0
webrtc: (vie_codec_impl.cc:518): DeregisterDecodeObserver for channel 0
webrtc: (vie_base_impl.cc:205): Channel deleted 0
Destroyed channel
Error(webrtcsession.cc:958): SetAudioSend: No audio channel exists.
Warning(webrtcsession.cc:996): Video not used in this call.
Warning(webrtcsession.cc:1029): SetVideoSend: No video channel exists.
Destroying NSS identity
Session:1051969493015781748 Old state:STATE_RECEIVEDTERMINATE New state:STATE_DEINIT Type:urn:xmpp:jingle:apps:rtp:1 Transport:http://www.google.com/transport/p2p
Camera 'com.apple.avfoundation.avcapturedevice.built-in_video:1' stopped after capturing 6 frames and dropping 0%
WebRtcVideoEngine::Terminate
webrtc: (vie_render_impl.cc:75): DeRegisterVideoRenderModule: 
webrtc: (vie_base_impl.cc:68): SetVoiceEngine: SetVoiceEngine
WebRtcVoiceEngine::Terminate
WebRtcVideoEngine::~WebRtcVideoEngine
webrtc: (vie_impl.cc:138): SetTraceCallback: 
2015-02-09 15:29:56.941 HelloCordova[1097:683042] sendMessage 1

And my code :

document.addEventListener("deviceready", onDeviceReady, false);


function onDeviceReady() {

    var session;

    var endButton = document.getElementById('endCall');
    endButton.addEventListener("click", function () {
                               session.bye();
                               alert("Call Ended");
                               }, false);

    var SIP = cordova.require("com.onsip.cordova.Sipjs");

    // Use plugin version of MediaHandler
    var PhoneRTCMediaHandler = cordova.require("com.onsip.cordova.SipjsMediaHandler")(SIP);

    // Set my UA to use the MediaHandler
    var ua = new SIP.UA({
                        mediaHandlerFactory: PhoneRTCMediaHandler,
                        authorizationUser : 'xxxxxxxx',
                        realm: 'xxxxx',
                        domain: '000.000.000.000',
                        password : 'xxxxxx',
                        wsServer : ['ws://000.000.000.000:8080/ws'],
                        displayName: 'xxxx',
                        registerExpires: 200
                        });

    var options = {
    media: {
    constraints: {
    audio: true,
    video: false
    },
    render: {
    remote: {
    audio: document.getElementById('remoteVideo')
    },
    local: {
    audio: document.getElementById('localVideo')
    }
    }
    }
    };
    //makes the call


    session = ua.invite('sip:00000000@xivo', options);


    console.log('UA');
}

Thank you for your helping :)

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