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Baresip is a portable and modular SIP User-Agent with audio and video support.
Copyright (c) 2010 - 2024 Alfred E. Heggestad and Contributors
Distributed under BSD license

Build Lint OpenSSL and LibreSSL Valgrind

Features:

  • Call features:

    • Unlimited number of SIP accounts
    • Unlimited number of calls
    • Unattended call transfer
    • Auto answer
    • Call hold and resume
    • Microphone mute
    • Call waiting
    • Call recording
    • Peer to peer calls
    • Video calls
    • Instant Messaging
    • Custom ring tones
    • Repeat last call (redial)
    • Message Waiting Indication (MWI)
    • Address book with presence
    • Conferencing
  • Signaling:

    • SIP protocol support
    • SIP outbound protocol for NAT-traversal
    • SIP Re-invite
    • SIP Routes
    • SIP early media support
    • DNS NAPTR/SRV support
    • Multiple accounts support
    • DTMF support (RTP, SIP INFO)
    • Multicast sending & receiving
  • Security:

    • Signalling encryption (TLS)
    • Audio and video encryption (Secure RTP)
    • DTLS-SRTP key exchange protocol
    • ZRTP key exchange protocol
    • SDES key exchange protocol
  • Audio:

    • Low latency audio pipeline
    • High definition audio codecs
    • Audio device configuration
    • Audio filter plugins
    • Internal audio resampler for fixed sampling rates
    • Linear 16 bit wave format support for ringtones
    • Packet loss concealment (PLC)
    • Configurable ringtone playback device
    • Automatic gain control (AGC) and Noise reducation
    • Acoustic echo control (AEC)
    • Configurable audio sample format (Signed 16-bit, 24-bit, Float etc)
    • EBU ACIP (Audio Contribution over IP) Profile
  • Audio-codecs:

    • AAC
    • aptX
    • AMR narrowband, AMR wideband
    • Codec2
    • G.711
    • G.722
    • G.726
    • L16
    • MPA
    • Opus
  • Audio-drivers:

    • Advanced Linux Sound Architecture (ALSA) audio-driver
    • PulseAudio POSIX OSes audio-driver
    • Android OpenSLES audio-driver
    • Gstreamer playbin input audio-driver
    • JACK Audio Connection Kit audio-driver
    • MacOSX/iOS coreaudio/audiounit audio-driver
    • Portaudio audio-driver
    • Windows winwave audio-driver
  • Video:

    • Support for H.264, H.265, VP8, VP9, AV1 Video
    • Configurable resolution/framerate/bitrate
    • Configurable video input/output
    • Support for asymmetric video
    • Configurable video pixel format
    • Hardware acceleration for video encoder/decoder
  • Video-codecs:

    • AV1
    • H.264
    • H.265
    • VP8
    • VP9
  • Video-drivers:

    • iOS avcapture video-source
    • FFmpeg/libav libavformat/avdevice input
    • Direct Show video-source
    • MacOSX AVCapture video-source
    • Linux V4L/V4L2 video-source
    • X11 grabber video-source
    • DirectFB video-output
    • SDL2 video-output
    • X11 video-output
  • NAT-traversal:

    • STUN support
    • TURN server support
    • ICE support
    • NATPMP support
    • PCP (Port Control Protocol) support
  • Networking:

    • multihoming, IPv4/IPv6
    • automatic network roaming
  • Management:

    • Embedded web-server with HTTP interface
    • Command-line console over UDP/TCP
    • Command line interface (CLI)
    • Simple configuration files
    • MQTT (Message Queue Telemetry Transport) module
  • Profiles:

    • EBU ACIP (Audio Contribution over IP) Profile

Building

baresip is using CMake, and the following packages must be installed before building:

See Wiki: Install Stable Release or Wiki: Install GIT Version for a full guide.

Build with debug enabled

$ cmake -B build
$ cmake --build build -j
$ cmake --install build

Build with release

$ cmake -B build -DCMAKE_BUILD_TYPE=Release 
$ cmake --build build -j

Build with selected modules

$ cmake -B build -DMODULES="menu;account;g711"
$ cmake --build build -j

Build with custom app modules

$ cmake -B build -DAPP_MODULES_DIR=../baresip-apps/modules -DAPP_MODULES="auloop;vidloop"
$ cmake --build build -j

Build with clang compiler

$ cmake -B build -DCMAKE_C_COMPILER=clang -DCMAKE_CXX_COMPILER=clang++
$ cmake --build build -j

Build static

$ cmake -B build -DSTATIC=ON
$ cmake --build build -j

Modules will be built if external dependencies are installed. After building you can start baresip like this:

$ build/baresip

The config files in $HOME/.baresip are automatically generated the first time you run baresip.

Build documentation

The API documentation can be build using doxygen.

$ doxygen mk/Doxyfile

By default the documentation is written to ../baresip-dox, if you want to change the destination directory you can change the OUTPUT_DIRECTORY in mk/Doxyfile.

Examples

  • Configuration examples are available in the examples directory.
  • Documentation on configuring baresip can be found in the Wiki.

License

The baresip project is using the 3-clause BSD license.

Contributing

Patches can be sent via Github Pull-Requests or to the Baresip mailing-list.

Design goals:

  • Minimalistic and modular VoIP client
  • SIP, SDP, RTP/RTCP, STUN/TURN/ICE
  • IPv4 and IPv6 support
  • RFC-compliancy
  • Robust, fast, low footprint
  • Portable C99 and C11 source code

Modular Plugin Architecture:

aac           Advanced Audio Coding (AAC) audio codec
account       Account loader
alsa          ALSA audio driver
amr           Adaptive Multi-Rate (AMR) audio codec
aptx          Audio Processing Technology codec (aptX)
aubridge      Audio bridge module
auconv        Audio sample format converter
audiounit     AudioUnit audio driver for MacOSX/iOS
aufile        Audio module for using a WAV-file as audio input
auresamp      Audio resampler
ausine        Audio sine wave input module
av1           AV1 video codec
avcapture     Video source using iOS AVFoundation video capture
avcodec       Video codec using FFmpeg/libav libavcodec
avfilter      Video filter using FFmpeg libavfilter
avformat      Video source using FFmpeg/libav libavformat
codec2        Codec2 low bit rate speech codec
cons          UDP/TCP console UI driver
contact       Contacts module
coreaudio     Apple macOS Coreaudio driver
ctrl_dbus     Control interface using DBUS
ctrl_tcp      TCP control interface using JSON payload
debug_cmd     Debug commands
directfb      DirectFB video display module
dshow         Windows DirectShow video source
dtls_srtp     DTLS-SRTP end-to-end encryption
ebuacip       EBU ACIP (Audio Contribution over IP) Profile
echo          Echo server module
evdev         Linux input driver
fakevideo     Fake video input/output driver
g711          G.711 audio codec
g722          G.722 audio codec
g7221         G.722.1 audio codec
g726          G.726 audio codec
gst           Gstreamer audio source
gtk           GTK+ 3 menu-based UI
gzrtp         ZRTP module using GNU ZRTP C++ library
httpd         HTTP webserver UI-module
httpreq       HTTP request module
ice           ICE protocol for NAT Traversal
jack          JACK Audio Connection Kit audio-driver
l16           L16 audio codec
menu          Interactive menu
mixausrc      Mixes another audio source into audio stream
mixminus      Mixes N-1 audio streams for conferencing
mpa           MPA Speech and Audio Codec
mqtt          MQTT (Message Queue Telemetry Transport) module
mwi           Message Waiting Indication
natpmp        NAT Port Mapping Protocol (NAT-PMP) module
netroam       Detects and applies changes of the local network addresses
opensles      OpenSLES audio driver
opus          OPUS Interactive audio codec
opus_multistream    OPUS multistream audio codec
pcp           Port Control Protocol (PCP) module
plc           Packet Loss Concealment (PLC) using spandsp
portaudio     Portaudio driver
pulse         Pulseaudio driver
presence      Presence module
rtcpsummary   RTCP summary module
sdl           Simple DirectMedia Layer 2.0 (SDL) video output driver
selfview      Video selfview module
serreg        Serial registration
snapshot      Save video-stream as PNG images
sndfile       Audio dumper using libsndfile
sndio         Audio driver for OpenBSD
srtp          Secure RTP encryption (SDES) using libre SRTP-stack
stdio         Standard input/output UI driver
stun          Session Traversal Utilities for NAT (STUN) module
swscale       Video scaling using libswscale
syslog        Syslog module
turn          Obtaining Relay Addresses from STUN (TURN) module
uuid          UUID generator and loader
v4l2          Video4Linux2 video source
vidbridge     Video bridge module
vidinfo       Video info overlay module
vp8           VP8 video codec
vp9           VP9 video codec
vumeter       Display audio levels in console
webrtc_aec    Acoustic Echo Cancellation (AEC) using WebRTC SDK
webrtc_aecm   Acoustic Echo Cancellation (AEC) mobile using WebRTC SDK
wincons       Console input driver for Windows
winwave       Audio driver for Windows
x11           X11 video output driver

IETF RFC/I-Ds:

  • RFC 2250 RTP Payload Format for the mpa Speech and Audio Codec

  • RFC 3016 RTP Payload Format for MPEG-4 Audio/Visual Streams

  • RFC 3262 Reliability of Provisional Responses for SIP

  • RFC 3311 SIP UPDATE Method

  • RFC 3428 SIP Extension for Instant Messaging

  • RFC 3711 The Secure Real-time Transport Protocol (SRTP)

  • RFC 3640 RTP Payload Format for Transport of MPEG-4 Elementary Streams

  • RFC 3856 A Presence Event Package for SIP

  • RFC 3863 Presence Information Data Format (PIDF)

  • RFC 3891 The SIP "Replaces" Header

  • RFC 4145 TCP-Based Media Transport in SDP

  • RFC 4240 Basic Network Media Services with SIP (partly)

  • RFC 4347 Datagram Transport Layer Security

  • RFC 4568 SDP Security Descriptions for Media Streams

  • RFC 4572 Connection-Oriented Media Transport over TLS Protocol in SDP

  • RFC 4574 The SDP Label Attribute

  • RFC 4585 Extended RTP Profile for RTCP-Based Feedback (RTP/AVPF)

  • RFC 4587 RTP Payload Format for H.261 Video Streams

  • RFC 4796 The SDP Content Attribute

  • RFC 4867 RTP Payload Format for the AMR and AMR-WB Audio Codecs

  • RFC 4961 Symmetric RTP / RTP Control Protocol (RTCP)

  • RFC 5285 A General Mechanism for RTP Header Extensions

  • RFC 5373 Requesting Answering Modes for SIP

  • RFC 5506 Support for Reduced-Size RTCP

  • RFC 5576 Source-Specific Media Attributes in SDP

  • RFC 5577 RTP Payload Format for ITU-T Recommendation G.722.1

  • RFC 5626 Managing Client-Initiated Connections in SIP

  • RFC 5627 Obtaining and Using GRUUs in SIP

  • RFC 5761 Multiplexing RTP Data and Control Packets on a Single Port

  • RFC 5763 Framework for Establishing a SRTP Security Context Using DTLS

  • RFC 5764 DTLS Extension to Establish Keys for SRTP

  • RFC 5888 The SDP Grouping Framework

  • RFC 6157 IPv6 Transition in SIP

  • RFC 6184 RTP Payload Format for H.264 Video

  • RFC 6263 App. Mechanism for Keeping Alive NAT Associated with RTP / RTCP

  • RFC 6416 RTP Payload Format for MPEG-4 Audio/Visual Streams

  • RFC 6464 A RTP Header Extension for Client-to-Mixer Audio Level Indication

  • RFC 6716 Definition of the Opus Audio Codec

  • RFC 6886 NAT Port Mapping Protocol (NAT-PMP)

  • RFC 7064 URI Scheme for STUN Protocol

  • RFC 7065 TURN Uniform Resource Identifiers

  • RFC 7310 RTP Payload Format for Standard apt-X and Enhanced apt-X Codecs

  • RFC 7587 RTP Payload Format for the Opus Speech and Audio Codec

  • RFC 7741 RTP Payload Format for VP8 Video

  • RFC 7742 WebRTC Video Processing and Codec Requirements

  • RFC 7798 RTP Payload Format for High Efficiency Video Coding (HEVC)

  • RFC 8285 A General Mechanism for RTP Header Extensions

  • RFC 8843 Negotiating Media Multiplexing Using SDP

  • draft-ietf-payload-vp9-16

Supported platforms:

  • Android (7.0 or later)
  • Apple macOS (10.12+)
  • Apple iOS 10.0 or later
  • Linux (kernel 4.0 or later, and glibc 2.5.x or later)
  • Windows 10 or later (mingw and VS2019)

Supported versions of C Standard library

  • Android bionic
  • BSD libc
  • GNU C Library (glibc)
  • Musl
  • Windows C Run-Time Libraries (CRT)
  • uClibc

Supported compilers:

  • gcc 9.x or later
  • MSVC 2019, 2022
  • clang 9.x or later

Supported versions of OpenSSL

  • OpenSSL version 1.1.1
  • OpenSSL version 3.x.x
  • LibreSSL version 3.x

Related projects

References

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baresip-ios's Issues

Can't get Audio.

@jfont555 thanks for getting back to me. Ended up getting sorted building the libraries locally actually and it turns out that my version of this problem was that I hadn't enabled the 'coreaudio' module under EXTRA_MODULES in the modules.mk file.

Essentially, I ended up building my own ObjC handler and passed it to baresip using uag_event_register() and ensuring that the library doesn't set up it's own call event handler over the top.

It seems to work pretty well though I'm having a couple of issues I'm experiencing currently with regards to catching certain events with it that I'm looking into.

Glad you got yours sorted. It has been a mission with little to no documentation.

I have same issue, can't get Audio, and I tried with your solution, like
In modules.mk I have changed code


#ifneq ($(USE_COREAUDIO),)

MODULES   += coreaudio

#endif

but now I am getting following error.
Screenshot-2019-04-25-at-6-05-28-PM

Error is :

error: use of undeclared identifier 'AudioObjectPropertyAddress'
AudioObjectPropertyAddress propertyAddress = {

Could you please help me for the same?

If I uncomment the above code and run the command, all working fine there is no issue, but can't get audio in call.

Thanks in advance...

Originally posted by @samirmagnates in #17 (comment)

31mua: no network

I've tried the taresip cocoa pod and it worked in the simulator but since it didn't work on a real device (arm64) I followed the guide here to compile the libraries myself but I think I might not have added them right to my project, now the code runs on the simulator but it writes this to the console right after starting.

[31mua: no network
�[;merror: stack

update: solved the no network error by adding
guard baresip_init(conf_config(), 0) == 0 else { print("baresip init error"); return }
to my code but now on the simulator it successfully establishes the call but I hear no audio, but when I run the same code on a real iPhone X it gives me a proxy error

code below:
`
Import UIKit

class ViewController: UIViewController {

override func viewDidLoad() {
    super.viewDidLoad()
    // Do any additional setup after loading the view.
    
}

override func viewWillAppear(_ animated: Bool) {
    var agent: OpaquePointer? = nil
    var client: SipClient? = nil
    do {
        client = try SipClient(agent: &agent)
    } catch {
        print("error: \(error) FDX")
    }
}

enum SipError: Error {
    case libre
    case config
    case stack
    case modules
    case userAgent
    case call
}

final class SipClient {
    
    required init?(agent: inout OpaquePointer?) throws {
        guard libre_init() == 0 else { throw SipError.libre }
        
        // Initialize dynamic modules.
        mod_init()
        print("after mod init")
        
        // Make configure file.
        
        if let path = NSSearchPathForDirectoriesInDomains(.documentDirectory, .userDomainMask, true).first {
            conf_path_set(path)
        }
        guard conf_configure() == 0 else { throw SipError.config}
        guard baresip_init(conf_config(), 0) == 0 else { print("baresip init error"); return }
        
        // Initialize the SIP stack.
        guard ua_init("baresip v" + BARESIP_VERSION  + " (" + ARCH + "/" + OS + ")", 1, 1, 1, 0) == 0 else { throw SipError.stack }
        
        // Load modules.
        guard conf_modules() == 0 else { throw SipError.modules }
        
        let addr = "sip:[email protected]:5060;transport=udp;answermode=auto;auth_pass=101;"
        
        // Start user agent.
        guard ua_alloc(&agent, addr) == 0 else { throw SipError.userAgent }
        
        // Make an outgoing call.
        
        guard ua_connect(agent, nil, nil, "sip:[email protected]:5060", VIDMODE_OFF) == 0 else { throw SipError.call }

        // Start the main loop.
        re_main(nil)
    }
    
    func close(agent: OpaquePointer) {
        mem_deref(UnsafeMutablePointer(agent))
        ua_close()
        mod_close()
    
        // Close and check for memory leaks.
        libre_close()
        tmr_debug()
        mem_debug()
    }  
}

}
`

install destination error

installation is supposed to be in /Users/user/git/baresip-ios/contrib/x86_64/ but all installation paths (including include files and libraries are appended a redundant path Users/user/git/baresip-ios/re/dist/. (user being the username)

...
installing /Users/user/git/baresip-ios/contrib/x86_64/Users/user/git/baresip-ios/re/dist/include/re/re_btrace.h
...

fatal error: /Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/lipo: can't open input file: /Users/user/git/baresip-ios/contrib/x86_64/lib/libre.a (No such file or directory)

No audio recording

Thank you for your work on baresip!
Is it possible to understand how it should work? It seems the audio module doesn't record the audio. How to check is it active at all?
Use baresip version 1.0 with EXTRA_MODULES='g711 srtp audiounit avcapture opus'

aucodec: PCMA/8000/1
...
avcapture: found video device 'Back Camera'
avcapture: found video device 'Front Camera'
medianat: stun
medianat: turn
medianat: ice
....
Populated 2 audio codecs
Populated 0 audio filters
Populated 0 video codecs
Populated 0 video filters
...
ua: using best effort AF: af=AF_INET
call: alloc with params laddr=<redacted>, af=AF_INET, use_rtp=1
call: use_video=0
call: connecting to <redacted>..
audio: start
...
call: SIP Progress: 100 Trying (/)
call: SIP Progress: 100 Giving it a try (/)
call: SIP Progress: 180 Ringing (/)
CALL_RINGING
....
call: got SDP answer (244 bytes)
call: update media
CALL_REMOTE_SDP
stream: update 'audio'
stream: audio: starting RTCP with remote <redacted>
audio: Set audio decoder: PCMA 8000Hz 1ch
audio: start
audio: Set audio encoder: PCMA 8000Hz 1ch
audio: start
audio tx pipeline:       (src) ---> PCMA
audio rx pipeline:      (play) <--- PCMA
call: stream start (active=1)
audio: start
audio: start
<redacted>: Call established: <redacted>
CALL_ESTABLISHED
[0:00:00] audio=0/0 (bit/s)    
[0:00:00] audio=0/0 (bit/s)    
[0:00:00] audio=0/0 (bit/s)    
[0:00:00] audio=0/0 (bit/s)    
[0:00:00] audio=0/0 (bit/s)    
[0:00:00] audio=0/0 (bit/s)    
[0:00:00] audio=0/0 (bit/s)    
[0:00:00] audio=0/0 (bit/s)    
[0:00:01] audio=0/0 (bit/s)    
.....
[0:00:13] audio=0/0 (bit/s)    
[0:00:13] audio=0/0 (bit/s)    
[0:00:13] audio=0/0 (bit/s)    
[0:00:13] audio=0/0 (bit/s)    
[0:00:14] audio=0/0 (bit/s)    
[0:00:14] audio=0/0 (bit/s)    
[0:00:14] audio=0/0 (bit/s) 
.....
<redacted>: session closed: Connection reset by peer
CALL_CLOSED
sip:bonjour@local: Call with <redacted> terminated (duration: 14 secs)
audio: destroyed (started=1)

Opus support?

Does this also include the opus libraries into the build?

Can't compile g722 codec

modules/g722/g722.c:13:10: fatal error: 'spandsp.h' file not found
I've installed spandsp with Homebrew like it says in the wiki but it still can't find the header
managed to fix this by adding a CPATH to /usr/local/include
but now during building in Xcode I get this error

Undefined symbols for architecture x86_64: "_g722_decode", referenced from: _decode in libbaresip.a(g722.o) "_g722_decode_init", referenced from: _decode_update in libbaresip.a(g722.o) "_g722_encode", referenced from: _encode in libbaresip.a(g722.o) "_g722_encode_init", referenced from: _encode_update in libbaresip.a(g722.o) ld: symbol(s) not found for architecture x86_64 clang: error: linker command failed with exit code 1 (use -v to see invocation)

Undefined symbols for architecture arm64: Can compile baresip but gives errors while using it in xcode project

I can compile baresip with modification in contrib file like below:
EXTRA_MODULES='g711 audiounit avformat avcapture opengles avcodec'

Terminal results:
LD libre.dylib
AR libre.a
LD libre.dylib
AR libre.a
LD libre.dylib
AR libre.a
LD libre.dylib
AR libre.a
LD librem.dylib
AR librem.a
LD librem.dylib
AR librem.a
LD librem.dylib
AR librem.a
LD librem.dylib
AR librem.a
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/audio.o
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/aulevel.o
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/conf.o
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/config.o
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/h264.o
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/play.o
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/video.o
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/vidutil.o
SH src/static.c
CC /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/src/static.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/g711/g711.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/audiounit/audiounit.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/audiounit/sess.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/audiounit/player.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/audiounit/recorder.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/avformat/avformat.o
OC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/avcapture/avcapture.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/opengles/opengles.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/avcodec/avcodec.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/avcodec/decode.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/avcodec/encode.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/avcodec/h263.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/auloop/auloop.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/fakevideo/fakevideo.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/selfview/selfview.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/vidbridge/vidbridge.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/vidbridge/src.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/vidbridge/disp.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/vidinfo/draw.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/vidinfo/vidinfo.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/vidloop/vidloop.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/vumeter/vumeter.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/aubridge/device.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/aubridge/src.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/aubridge/play.o
CC [m] /Users/xxx/Desktop/baresip-ios-master/build/aarch64/baresip/modules/aufile/aufile.o
AR libbaresip.a

But now when I use these file Contrib > aarch64 to my project it gives error while build like :

Undefined symbols for architecture arm64:
"_av_opt_set_defaults", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_opt_set_int", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_hwframe_ctx_alloc", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_hwframe_ctx_init", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_hwframe_get_buffer", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_opt_set", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_frame_copy_props", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_packet_alloc", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_avcodec_receive_packet", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_packet_free", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_avcodec_free_context", referenced from:
_destructor in libbaresip.a(decode.o)
_destructor in libbaresip.a(encode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_avcodec_send_packet", referenced from:
_read_thread in libbaresip.a(avformat.o)
_ffdecode in libbaresip.a(decode.o)
"_av_seek_frame", referenced from:
_read_thread in libbaresip.a(avformat.o)
"_avformat_open_input", referenced from:
_alloc in libbaresip.a(avformat.o)
"_avformat_find_stream_info", referenced from:
_alloc in libbaresip.a(avformat.o)
"_av_free", referenced from:
_destructor in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_avformat_network_init", referenced from:
_module_init in libbaresip.a(avformat.o)
"_avcodec_parameters_to_context", referenced from:
_alloc in libbaresip.a(avformat.o)
"_avcodec_open2", referenced from:
_alloc in libbaresip.a(avformat.o)
_avcodec_decode_update in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_avcodec_send_frame", referenced from:
_avcodec_encode in libbaresip.a(encode.o)
"_av_hwframe_transfer_data", referenced from:
_ffdecode in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_av_frame_alloc", referenced from:
_read_thread in libbaresip.a(avformat.o)
_avcodec_decode_update in libbaresip.a(decode.o)
_ffdecode in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_avcodec_close", referenced from:
_destructor in libbaresip.a(avformat.o)
"_av_read_frame", referenced from:
_read_thread in libbaresip.a(avformat.o)
"_avcodec_alloc_context3", referenced from:
_alloc in libbaresip.a(avformat.o)
_avcodec_decode_update in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_avformat_network_deinit", referenced from:
_module_close in libbaresip.a(avformat.o)
"_avcodec_find_encoder", referenced from:
_avcodec_encode_update in libbaresip.a(encode.o)
"_avcodec_find_encoder_by_name", referenced from:
_module_init in libbaresip.a(avcodec.o)
_avcodec_encode in libbaresip.a(encode.o)
"_avcodec_find_decoder_by_name", referenced from:
_module_init in libbaresip.a(avcodec.o)
"_avcodec_find_decoder", referenced from:
_module_init in libbaresip.a(avcodec.o)
_alloc in libbaresip.a(avformat.o)
_avcodec_decode_update in libbaresip.a(decode.o)
"_av_frame_free", referenced from:
_read_thread in libbaresip.a(avformat.o)
_ffdecode in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_av_get_pix_fmt_name", referenced from:
_module_init in libbaresip.a(avcodec.o)
_read_thread in libbaresip.a(avformat.o)
_ffdecode in libbaresip.a(decode.o)
"_av_buffer_unref", referenced from:
_module_close in libbaresip.a(avcodec.o)
_avcodec_encode in libbaresip.a(encode.o)
"_av_hwdevice_find_type_by_name", referenced from:
_module_init in libbaresip.a(avcodec.o)
"_avcodec_receive_frame", referenced from:
_read_thread in libbaresip.a(avformat.o)
_ffdecode in libbaresip.a(decode.o)
"_avdevice_register_all", referenced from:
_module_init in libbaresip.a(avformat.o)
"_avformat_close_input", referenced from:
_destructor in libbaresip.a(avformat.o)
"_av_buffer_ref", referenced from:
_avcodec_decode_update in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_av_strerror", referenced from:
_module_init in libbaresip.a(avcodec.o)
_ffdecode in libbaresip.a(decode.o)
_avcodec_encode in libbaresip.a(encode.o)
"_av_hwdevice_get_type_name", referenced from:
_module_init in libbaresip.a(avcodec.o)
_avcodec_decode_update in libbaresip.a(decode.o)
"_avcodec_get_hw_config", referenced from:
_module_init in libbaresip.a(avcodec.o)
"_av_packet_unref", referenced from:
_read_thread in libbaresip.a(avformat.o)
"_av_init_packet", referenced from:
_read_thread in libbaresip.a(avformat.o)
_ffdecode in libbaresip.a(decode.o)
"_av_hwdevice_ctx_create", referenced from:
_module_init in libbaresip.a(avcodec.o)
ld: symbol(s) not found for architecture arm64
clang: error: linker command failed with exit code 1 (use -v to see invocation)

Don't know why? Thanks in advance

Audio issue in baresip in iOS

I checked this baresip/baresip#1174
but it's not helped me

I can't get audio from both sides.

  • full logs:

--- baresip_init_result 0
ua: adding SIP CA file: /private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/cert.pem
ua: adding SIP CA path:
--- ua_init_result 0
�[31mdl: mod: ./stdio.so (dlopen(./stdio.so, 0x0006): tried: '/usr/lib/system/introspection/stdio.so' (no such file, not in dyld cache), './stdio.so' (no such file), '/private/preboot/Cryptexes/OS./stdio.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./stdio.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./stdio.so' (no such file), '/usr/lib/./stdio.so' (no such file, not in dyld cache), './stdio.so' (no such file), '/usr/lib/system/introspection/stdio.so' (no such file, not in dyld cache), '//stdio.so' (no such file), '/private/preboot/Cryptexes/OS//stdio.so' (no such file), '//stdio.so' (no such file))
�[;m�[31mmodule stdio.so: No such file or directory
�[;maucodec: PCMU/8000/1
aucodec: PCMA/8000/1
aufilt: auconv
aufilt: auresamp
auplay: audiounit
ausrc: audiounit
vidsrc: avcapture
medianat: stun
medianat: turn
medianat: ice
Populated 0 accounts
account: No SIP accounts found
-- check your config or add an account using 'uanew' command
Populated 3 contacts
�[31mdl: mod: ./netroam.so (dlopen(./netroam.so, 0x0006): tried: '/usr/lib/system/introspection/netroam.so' (no such file, not in dyld cache), './netroam.so' (no such file), '/private/preboot/Cryptexes/OS./netroam.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./netroam.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./netroam.so' (no such file), '/usr/lib/./netroam.so' (no such file, not in dyld cache), './netroam.so' (no such file), '/usr/lib/system/introspection/netroam.so' (no such file, not in dyld cache), '//netroam.so' (no such file), '/private/preboot/Cryptexes/OS//netroam.so' (no such file), '//netroam.so' (no such file))
�[;m�[31mmodule netroam.so: No such file or directory
�[;mPopulated 2 audio codecs
Populated 2 audio filters
Populated 0 video codecs
Populated 0 video filters
2023-10-03 13:45:10.410411+0300 Test5[2378:666716] --- conf_modules_result 0
2023-10-03 13:45:10.410472+0300 Test5[2378:666716] --- uag_event_register_result 0
�[31mdl: mod: ./stdio.so (dlopen(./stdio.so, 0x0006): tried: '/usr/lib/system/introspection/stdio.so' (no such file, not in dyld cache), './stdio.so' (no such file), '/private/preboot/Cryptexes/OS./stdio.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./stdio.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./stdio.so' (no such file), '/usr/lib/./stdio.so' (no such file, not in dyld cache), './stdio.so' (no such file), '/usr/lib/system/introspection/stdio.so' (no such file, not in dyld cache), '//stdio.so' (no such file), '/private/preboot/Cryptexes/OS//stdio.so' (no such file), '//stdio.so' (no such file))
�[;m�[31mmodule stdio.so: No such file or directory
�[;mstatic module already loaded: g711.so
static module already loaded: auconv.so
static module already loaded: auresamp.so
static module already loaded: audiounit.so
static module already loaded: avcapture.so
static module already loaded: uuid.so
static module already loaded: stun.so
static module already loaded: turn.so
static module already loaded: ice.so
static module already loaded: account.so
static module already loaded: contact.so
static module already loaded: debug_cmd.so
static module already loaded: menu.so
�[31mdl: mod: ./netroam.so (dlopen(./netroam.so, 0x0006): tried: '/usr/lib/system/introspection/netroam.so' (no such file, not in dyld cache), './netroam.so' (no such file), '/private/preboot/Cryptexes/OS./netroam.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./netroam.so' (no such file), '/private/var/containers/Bundle/Application/3D87F0EF-C40E-4975-9F08-C582DD204B3B/Test5.app/Frameworks/./netroam.so' (no such file), '/usr/lib/./netroam.so' (no such file, not in dyld cache), './netroam.so' (no such file), '/usr/lib/system/introspection/netroam.so' (no such file, not in dyld cache), '//netroam.so' (no such file), '/private/preboot/Cryptexes/OS//netroam.so' (no such file), '//netroam.so' (no such file))
�[;m�[31mmodule netroam.so: No such file or directory
�[;mPopulated 2 audio codecs
Populated 2 audio filters
Populated 0 video codecs
Populated 0 video filters
�[31mrand: rand_u32: random not inited
�[;musing stunserver: 'stun:voice.example.com'
2023-10-03 13:45:10.412405+0300 Test5[2378:666716] --- account_set_auth_user_result 0
2023-10-03 13:45:10.412445+0300 Test5[2378:666716] --- account_set_auth_pass_result 0
2023-10-03 13:45:10.412467+0300 Test5[2378:666716] --- account_set_outbound_result 0
2023-10-03 13:45:10.412487+0300 Test5[2378:666716] --- account_set_sipnat_result 0
�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;mcall: connecting to 'sip:@voice.example.com'..
�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m2023-10-03 13:45:10.413004+0300 Test5[2378:666716] ua_event: CALL_OUTGOING
�[31mrand: rand_u64: random not inited
�[;m2023-10-03 13:45:10.413271+0300 Test5[2378:666716] ua_event: CALL_LOCAL_SDP
2023-10-03 13:45:10.413299+0300 Test5[2378:666716] --- ua_connect_result 0
�[;mwebsock: connecting to 'wss://[
]:443/'
websock_connect: No route to host
ws_conn_send failed (No route to host)
�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;mwebsock: connecting to 'wss://*******:443/'

<0x10313b2c8> WSS websock established to :443
--> send
�[31mrand: rand_u64: random not inited
�[;mwebsock: connecting to 'wss://[
]:443/'
websock_connect: No route to host
ws_conn_send failed (No route to host)
�[;mcall: SIP Progress: 100 trying -- your call is important to us (/)

�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;mcall: SIP Progress: 183 Session Progress (application/sdp)
[email protected]: Call in-progress: sip:*****@voice.example.com
2023-10-03 13:45:22.967734+0300 Test5[2378:666716] ua_event: CALL_PROGRESS
2023-10-03 13:45:22.967902+0300 Test5[2378:666716] ua_event: CALL_REMOTE_SDP
stream: update 'audio'
�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;mcall: SIP Progress: 183 Session Progress (application/sdp)
[email protected]: Call in-progress: sip:****@voice.example.com
2023-10-03 13:45:22.969137+0300 Test5[2378:666716] ua_event: CALL_PROGRESS
2023-10-03 13:45:22.969245+0300 Test5[2378:666716] ua_event: CALL_REMOTE_SDP
stream: update 'audio'

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)
�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;mcall: SIP Progress: 180 Ringing (application/sdp)
[email protected]: Call in-progress: sip:****@voice.example.com
ua_event: CALL_PROGRESS
ua_event: CALL_REMOTE_SDP
stream: update 'audio'

�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;[email protected]: Call answered: sip:@voice.example.com
ua_event: CALL_ANSWERED
ua_event: CALL_REMOTE_SDP
stream: update 'audio'
�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;[email protected]: Call established: sip:
@voice.example.com
2023-10-03 13:45:27.579481+0300 Test5[2378:666716] ua_event: CALL_ESTABLISHED

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:00] audio=0/0 (bit/s)

[0:00:01] audio=0/0 (bit/s)
[0:00:02] audio=0/0 (bit/s)

�[31mrand: rand_u64: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;m�[31mrand: rand_u32: random not inited
�[;msip:*****@voice.example.com: session closed: Connection reset by peer
ua_event: CALL_CLOSED
sip:[email protected]: Call with sip:****@voice.example.com terminated (duration: 13 secs)

No TLS transport on iOS ?

Hello all,

I have stumbled on yet another network connection issue on iOS with baresip. It seems that on iOS Baresip fails to bind the TLS socket. For example if checking 'SIP Debug' in menu on Android i get UDP, TCP and TLS transports. On iOS I'm getting only UDP and TCP. When I try to register an UA using TLS I get a error printout from Baresip saying something like: SIP register failed - transport not available.

Any suggestions on how to debug this and find the problem?

Br,
Per Enstedt - wx3 telecom

Make contrib error

This is the error that occurs when "make contrib". Has anyone come across this?

src/audio.c:422:10: error: implicit declaration of function 'aulevel_calc_dbov' 
is invalid in C99 [-Werror,-Wimplicit-function-declaration]
level = aulevel_calc_dbov(fmt, sampv, sampc); 
        ^

I can't hear voice of incoming call, but outgoing call works fine

If I have a outgoing audio call I have this log:

stream: 'audio' mnat 'ice' connected: raddr 145.239.65.46:14526 145.239.65.46:14527
stream: audio: starting mediaenc 'srtp' (wait_secure=0)
srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80)
call: mediaenc event 'Secure' (audio,AES_CM_128_HMAC_SHA1_80)

hear voice...

if incoming I have only this part:

stream: 'audio' mnat 'ice' connected: raddr 145.239.65.46:14538 145.239.65.46:14539
stream: audio: starting mediaenc 'srtp' (wait_secure=0)

without this part:

srtp: audio: SRTP is Enabled (cryptosuite=AES_CM_128_HMAC_SHA1_80)
call: mediaenc event 'Secure' (audio,AES_CM_128_HMAC_SHA1_80)

and I can't hear voice...

looks like something with srtp encryption side , how I can fix it ?
I have same configuration for both calls ...

tls-support ios

Has any one got tls working with ios? All I get is unsupported protocol. I've tried changing the USE_OPENSSL flag in contrib witout luck.

Cocoapods support

Is there any way you can provide a cocoapod for this project? This would make it way easier to stay up to date.

existing swift integration example

hello guys, do you know about any existing open source app solution using baresip-ios? It is hard to start without documenation, best would be to see existing iOS application integrated with baresip-ios

Error Code:43 Protocol not supported

I've been trying to solve this out.
I've been following this open source project which is written c/c++, I was trying to replicate the login/registration on sip server part, but always getting

SIP register failed: Protocol not supported

This is the code which I've tried so far.

Any insight to this issue?

AudioUnit hell

Hello,

So i have a big issue. The story starts like this: I need audio only (no video), a few days ago I tried using https://github.com/miche-atucha/taresip which was lovely managed to call, could hear the other client (even tho it had a small issue of unintentional audio passthrough, it was pretty much ok). ALSO, IT USED COREAUDIO surprised pikachu. As far as I looked, that was built with baresip-0.4.20, re-0.4.17 and rem-0.4.7.

Now here comes problem#1: I needed TLS. So i found this https://github.com/rtcexpert/baresip-ios-bitcode (bless his soul), which is also a pending PR here and it connects fine. So i took the mk, i grabbed baresip 0.6.6, re 0.6.1 and rem 0.6.0, openssl&co, built it just fine. problem is if i wanna add coreaudio module in

BARESIP_BUILD_FLAGS_X86_64 := \
	$(BARESIP_BUILD_FLAGS) \
	EXTRA_MODULES='g711 coreaudio srtp audiounit avcapture opengles'

it will fail with a metric ton of errors like here (baresip/baresip#593). So I did what @eriksundin suggested and patched coreaudio.c with

CFStringRef coreaudio_get_device_uid(const char *name)
{
	#if TARGET_OS_IPHONE
		return NULL;
	# else
        ....
       #endif
}

and then i got some more errors like

CC [m]  /Users/dani/Downloads/baresip-ios-bitcode-master/build/aarch64/baresip/modules/coreaudio/recorder.o
modules/coreaudio/recorder.c:78:51: error: too many arguments to function call, expected 2, have 3
        rh(inQB->mAudioData, inQB->mAudioDataByteSize/2, arg);

So I just gave up...

Now you ask what happens if i just use the default audiounit ? I get the following logs

2020-07-15 19:34:47.336010+0300 baresiptest[1728:295101] uaInitErr: 0
aucodec: PCMU/8000/1
aucodec: PCMA/8000/1
auplay: audiounit
ausrc: audiounit
medianat: stun
medianat: turn
medianat: ice
medianat: ice-lite
Populated 2 audio codecs
Populated 0 audio filters
Populated 0 video codecs
Populated 0 video filters
2020-07-15 19:34:47.378361+0300 baresiptest[1728:295101] confModulesErr: 0
2020-07-15 19:34:47.378498+0300 baresiptest[1728:295101] uaAllocErr: 0
2020-07-15 19:34:47.378883+0300 baresiptest[1728:295101] REGISTERED: YESSSSS
ua: using best effort AF: af=AF_INET
...........connecting.................
call: answering call from sip:2000@xxx with 200
stream: update 'audio'
audio: Set audio decoder: PCMU 8000Hz 1ch
audiounit: AudioRouteChange - reason 3
audio: player started with sample format S16LE
audio: Set audio encoder: PCMU 8000Hz 1ch
audiounit: record: enable resampler 8000.0 -> 8000 Hz
2020-07-15 19:34:49.054507+0300 baresiptest[1728:295140] [aid] AudioIssueDetectorNode.cpp:166:Initialize: Caught analyzer graph exception 1718775073 !mrf slice duration must be set before configure. in /Library/Caches/com.apple.xbs/Sources/AudioDSP/AudioDSP-366.38/CoreAudioUtility/Source/CADSP/DSPGraph/DSPGraph_Graph.cpp:628
audio: source started with sample format S16LE
audio tx pipeline:   audiounit ---> PCMU
audio rx pipeline:   audiounit <--- PCMU
�[31mmain: long async blocking: 656>100 ms (h=0x1046e3f48 arg=0x28152cbf0)
�[;mcall: SIP Progress: 180 Ringing (/)
call: SIP Progress: 180 Ringing (/)
2020-07-15 19:34:49.089235+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.112152+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.135369+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.158808+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.182018+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2000@xxx: Call established: sip:2000@xxx
2020-07-15 19:34:49.205226+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.228837+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.252263+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.275400+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.299691+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
stream: update 'audio'
audio: Set audio decoder: PCMU 8000Hz 1ch
2020-07-15 19:34:49.322460+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.344792+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
audio: player started with sample format S16LE
audio: Set audio encoder: PCMU 8000Hz 1ch
2020-07-15 19:34:49.682740+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.817741+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.821564+0300 baresiptest[1728:295140] [aurioc] AURemoteIO.cpp:1086:Initialize: failed: -66635 (enable 3, outf< 1 ch,      0 Hz, Float32> inf< 1 ch,   8000 Hz, Int16>)
�[31maudiounit: record failed: -66635 (\377\376\373\265)
�[;m�[31maudio: start_source failed (audiounit.default): Operation not supported by device
�[;msip:2000@xxx:5061;transport=tls: session closed: Operation not supported by device
2020-07-15 19:34:49.840826+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
�[31mmain: long async blocking: 531>100 ms (h=0x1046e3f48 arg=0x28152cbf0)
�[;m2020-07-15 19:34:49.864077+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.887167+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.910383+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.933624+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.956830+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:49.980111+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.003410+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.026485+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.049776+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.073054+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.096461+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.119868+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.143138+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.166310+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.189574+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.212550+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
2020-07-15 19:34:50.236056+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
sip:2000@xxx: session closed: Connection reset by peer
2020-07-15 19:34:50.258984+0300 baresiptest[1728:295168] AUBuffer.h:61:GetBufferList: EXCEPTION (-1) [mPtrState == kPtrsInvalid is false]: ""
sip:[email protected]:57128;transport=tls: Call with sip:2000@xxx terminated (duration: 1 sec)

I tried multiple configs such as:
https://gist.github.com/LonestarX91/ff4922fd03b9aa88a1e8dbbb79d78a32
https://gist.github.com/LonestarX91/483bfeb7327af18e5776dcf084d47770
same result
And obviously... no audio.

If anyone can please help me get either (coreaudio or audiounit) working, i'll raise them a statue

THANKS

libcrypto.dylib for architecture x86_64

Hi,
I want using this library for iOS 12 (64 bit)

When I run command:
$ make contrib

have error:

ld: building for iOS Simulator, but linking in dylib file (/usr/local/opt/openssl/lib/libcrypto.dylib) built for macOS, file '/usr/local/opt/openssl/lib/libcrypto.dylib' for architecture x86_64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
make[1]: *** [librem.dylib] Error 1
make: *** [librem] Error 2

Can your help me resolve it?

Problem compiling Baresip for IOS

  1. I am using "re", "rem" & "baresip" from Repository not from http://www.creytiv.com/pub
  2. I ran "make contrib"

Result:

`
LD librem.dylib
ld: warning: directory not found for option '-L-lre'
Undefined symbols for architecture arm64:
"_list_append", referenced from:
_aubuf_append in aubuf.o
_aumix_source_enable in aumix.o
_vidmix_source_enable in vidmix.o
"_list_count", referenced from:
_aumix_source_count in aumix.o
"_list_flush", referenced from:
_aubuf_destructor in aubuf.o
_aubuf_flush in aubuf.o
"_list_unlink", referenced from:
_auframe_destructor in aubuf.o
_source_destructor in aumix.o
_aumix_source_enable in aumix.o
_source_destructor in vidmix.o
_vidmix_source_enable in vidmix.o
"_lock_alloc", referenced from:
_aubuf_alloc in aubuf.o
"_lock_read_get", referenced from:
_aubuf_debug in aubuf.o
_aubuf_cur_size in aubuf.o
"_lock_rel", referenced from:
_aubuf_append in aubuf.o
_aubuf_read in aubuf.o
_aubuf_get in aubuf.o
_aubuf_flush in aubuf.o
_aubuf_debug in aubuf.o
_aubuf_cur_size in aubuf.o
"_lock_write_get", referenced from:
_aubuf_append in aubuf.o
_aubuf_read in aubuf.o
_aubuf_get in aubuf.o
_aubuf_flush in aubuf.o
"_mbuf_alloc", referenced from:
_aubuf_write in aubuf.o
"_mbuf_read_mem", referenced from:
_aubuf_read in aubuf.o
_avc_config_decode in config.o
"_mbuf_read_u16", referenced from:
_avc_config_decode in config.o
"_mbuf_read_u8", referenced from:
_avc_config_decode in config.o
_h264_nal_header_decode in nal.o
"_mbuf_write_mem", referenced from:
_aubuf_write in aubuf.o
_avc_config_encode in config.o
"_mbuf_write_u16", referenced from:
_autone_sine in tone.o
_autone_dtmf in tone.o
_avc_config_encode in config.o
"_mbuf_write_u8", referenced from:
_avc_config_encode in config.o
_h264_nal_header_encode in nal.o
"_mem_alloc", referenced from:
_aumix_thread in aumix.o
_aumix_source_alloc in aumix.o
"_mem_deref", referenced from:
_aubuf_alloc in aubuf.o
_aubuf_destructor in aubuf.o
_aubuf_append in aubuf.o
_auframe_destructor in aubuf.o
_aubuf_write in aubuf.o
_aubuf_read in aubuf.o
_aufile_open in aufile.o
...
"_mem_ref", referenced from:
_aubuf_append in aubuf.o
_aumix_source_alloc in aumix.o
_vidmix_source_alloc in vidmix.o
"_mem_zalloc", referenced from:
_aubuf_alloc in aubuf.o
_aubuf_append in aubuf.o
_aufile_open in aufile.o
_dtmf_dec_alloc in dec.o
_aumix_alloc in aumix.o
_aumix_thread in aumix.o
_aumix_source_alloc in aumix.o
...
"_re_fprintf", referenced from:
_wav_header_decode in wave.o
_auconv_from_s16 in auconv.o
_auconv_to_s16 in auconv.o
_vidframe_draw_point in draw.o
_vidframe_draw_hline in draw.o
_get_bit in getbit.o
_get_ue_golomb in getbit.o
...
"_re_hprintf", referenced from:
_aubuf_debug in aubuf.o
"_re_printf", referenced from:
_vidframe_init_buf in frame.o
_vidframe_fill in frame.o
_vidframe_copy in frame.o
_vidconv in vconv.o
"_sys_htoll", referenced from:
_wav_header_encode in wave.o
"_sys_htols", referenced from:
_wav_header_encode in wave.o
"_sys_ltohl", referenced from:
_wav_header_decode in wave.o
"_sys_ltohs", referenced from:
_wav_header_decode in wave.o
"_tmr_jiffies", referenced from:
_aubuf_get in aubuf.o
_aumix_thread in aumix.o
_content_thread in vidmix.o
_vidmix_thread in vidmix.o
ld: symbol(s) not found for architecture arm64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
make[1]: *** [librem.dylib] Error 1
make: *** [librem] Error 2

`

Note: Similar Error when compiling from http://www.creytiv.com/pub

Undefined symbols for Architecture arm64

Hi,

I was trying to build the latest version of Baresip (0.5.0), I'm also using libre-0.5.0 and librem-0.5.0, and I'm getting the following error:

  LD      libre.dylib
  AR      libre.a
  LD      libre.dylib
  AR      libre.a
  LD      libre.dylib
  AR      libre.a
  LD      libre.dylib
  AR      libre.a
  LD      libre.dylib
  AR      libre.a
  LD      librem.dylib
ld: warning: directory not found for option '-L-lre'
Undefined symbols for architecture arm64:
  "_list_append", referenced from:
      _aubuf_append in aubuf.o
      _aumix_source_enable in aumix.o
      _vidmix_source_enable in vidmix.o
  "_list_count", referenced from:
      _aumix_source_count in aumix.o
  "_list_flush", referenced from:
      _aubuf_destructor in aubuf.o
      _aubuf_flush in aubuf.o
  "_list_unlink", referenced from:
      _auframe_destructor in aubuf.o
      _source_destructor in aumix.o
      _aumix_source_enable in aumix.o
      _source_destructor in vidmix.o
      _vidmix_source_enable in vidmix.o
  "_lock_alloc", referenced from:
      _aubuf_alloc in aubuf.o
  "_lock_read_get", referenced from:
      _aubuf_debug in aubuf.o
      _aubuf_cur_size in aubuf.o
  "_lock_rel", referenced from:
      _aubuf_append in aubuf.o
      _aubuf_read in aubuf.o
      _aubuf_get in aubuf.o
      _aubuf_flush in aubuf.o
      _aubuf_debug in aubuf.o
      _aubuf_cur_size in aubuf.o
  "_lock_write_get", referenced from:
      _aubuf_append in aubuf.o
      _aubuf_read in aubuf.o
      _aubuf_get in aubuf.o
      _aubuf_flush in aubuf.o
  "_mbuf_alloc", referenced from:
      _aubuf_write in aubuf.o
  "_mbuf_read_mem", referenced from:
      _aubuf_read in aubuf.o
  "_mbuf_write_mem", referenced from:
      _aubuf_write in aubuf.o
  "_mbuf_write_u16", referenced from:
      _autone_sine in tone.o
      _autone_dtmf in tone.o
  "_mem_alloc", referenced from:
      _aumix_thread in aumix.o
      _aumix_source_alloc in aumix.o
  "_mem_deref", referenced from:
      _aubuf_alloc in aubuf.o
      _aubuf_destructor in aubuf.o
      _aubuf_append in aubuf.o
      _auframe_destructor in aubuf.o
      _aubuf_write in aubuf.o
      _aubuf_read in aubuf.o
      _aufile_open in aufile.o
      ...
  "_mem_ref", referenced from:
      _aubuf_append in aubuf.o
      _aumix_source_alloc in aumix.o
      _vidmix_source_alloc in vidmix.o
  "_mem_zalloc", referenced from:
      _aubuf_alloc in aubuf.o
      _aubuf_append in aubuf.o
      _aufile_open in aufile.o
      _aumix_alloc in aumix.o
      _aumix_thread in aumix.o
      _aumix_source_alloc in aumix.o
      _vidmix_alloc in vidmix.o
      ...
  "_re_fprintf", referenced from:
      _wav_header_decode in wave.o
      _auconv_from_s16 in auconv.o
      _auconv_to_s16 in auconv.o
      _vidframe_draw_point in draw.o
  "_re_hprintf", referenced from:
      _aubuf_debug in aubuf.o
  "_re_printf", referenced from:
      _vidframe_init_buf in frame.o
      _vidframe_fill in frame.o
      _vidframe_copy in frame.o
      _vidconv in vconv.o
  "_sys_htoll", referenced from:
      _wav_header_encode in wave.o
  "_sys_htols", referenced from:
      _wav_header_encode in wave.o
  "_sys_ltohl", referenced from:
      _wav_header_decode in wave.o
  "_sys_ltohs", referenced from:
      _wav_header_decode in wave.o
  "_tmr_jiffies", referenced from:
      _aubuf_get in aubuf.o
      _aumix_thread in aumix.o
      _content_thread in vidmix.o
      _vidmix_thread in vidmix.o
ld: symbol(s) not found for architecture arm64
clang: error: linker command failed with exit code 1 (use -v to see invocation)
make[1]: *** [librem.dylib] Error 1
make: *** [librem] Error 2

Any ideia on what might be causing this?
I'm trying to build on my OSX Machine, running MacOS 10.11.6

Please let me know if there is any other log or information about my machine that you might need.

Does Baresip/Libre run on Apple arm64?

Hi there,

I am trying to do sip registration on an iOS app using libre. I compiled the library to run on the following architectures: i386, armv7 and aarch64 (arm64). The app is built successfully on all architectures, and it does the sip registration on i386 (iphone simulator) and armv7. However, on arm64 everything runs normally until the line where the register sip packet should be sent it executes the statement but I don't see anything on wireshark and it waits for a reply but it gets nothing so an operation time out error appears. Has anyone tried running baresip or libre on an arm64 (iphone 5s, iphone 6 or iphone 6 plus)?

Thanks,
Taha

Install baresip 1.0.0 getting error of No rule to make target

I am trying to install updated baresip version 1.0.0, before that I alreday installed 0.6.5 and its successfuly installd: Now I get 4 zip from below link (master)

https://github.com/creytiv/rem
https://github.com/baresip/re
https://github.com/baresip/baresip
and
https://github.com/baresip/baresip-ios

added three unzip folder rem, re and baresip in baresip-ios-master

then fire command :
$ make contrib

its giving error at time of creating folder x86_64

CC /Users/samir/Downloads/Baresip1.0.0/baresip-ios-master/build/x86_64/librem/h264/nal.o CC /Users/samir/Downloads/Baresip1.0.0/baresip-ios-master/build/x86_64/librem/h264/sps.o LD librem.dylib AR librem.a Makefile:134: modules/opengles/module.mk: No such file or directory make[1]: *** No rule to make target modules/opengles/module.mk. Stop. make: *** [baresip] Error 2

I think its an issue of https://github.com/baresip/baresip-ios which is not updated as per baresip verison 1.0.0

architecture error

fatal error: /Applications/Xcode.app/Contents/Developer/Toolchains/XcodeDefault.xctoolchain/usr/bin/lipo: specifed architecture type (arm64) for file (/Users/Desktop/baresip-ios/contrib/aarch64/lib/libre.a) does not match its cputype (16777223) and cpusubtype (3) (should be cputype (16777228) and cpusubtype (0)) make: *** [libre] Error 1

am faceing this error!
what i do next?
Please help me!...

ua_init Error 49

Hello!!

In iOS when connected to wifi ua_init method returning 49 Error code but it's working fine in mobile data on the same device. let me know if you need any debug print. Kindly help me with this.

as mentioned in this issue, I have copied the route.h file in iOS SDK also

OsType: iOS
Model: iPhone 8 Plus
OsVersion: 14.4
Baresip v 1.1.0

`net_debug_log(): 36 - net_debug_log = --- Network debug ---
Local IPv4: [E] en0|192.168.18.12
Local IPv6: [E] en0|fe80::ca9:5c79:32e4:bf56
net interfaces:
lo0: 127.0.0.1
lo0: ::1
lo0: fe80::1
pdp_ip1: fe80::14bb:55f4:4c6b:2898
pdp_ip1: 2409:4032:6d06:c3d4:9e:2384:da50:a9dc
pdp_ip1: 2409:4032:6d06:c3d4:7084:a35c:477:914e
pdp_ip0: 26.60.29.192
pdp_ip0: fe80::8c2:2fa7:36a4:3ea8
pdp_ip0: 2409:4072:6d00:80aa:805:c8ae:9efc:f302
pdp_ip0: 2409:4072:6d00:80aa:8dd0:98b0:71c:8dd
en0: fe80::ca9:5c79:32e4:bf56
en0: 192.168.18.12
awdl0: fe80::249e:93ff:fe0f:ed79
utun0: fe80::e914:9594:a020:2149
utun1: fe80::1aaa:fa59:469a:f52
ipsec2: fe80::fa95:eaff:fea3:a04c
ipsec2: 2409:4032:6d06:c3d4:9e:2384:da50:a9dc
ipsec3: fe80::fa95:eaff:fea3:a04c
ipsec3: 2409:4032:6d06:c3d4:9e:2384:da50:a9dc
llw0: fe80::249e:93ff:fe0f:ed79
DNS Servers from Config: (1)
0: 192.168.18.1:53

20:38:00 20-05-2021 void net_dns_debug_log(): 90 - net_dns_debug_log = DNS Servers from Config: (1)
0: 192.168.18.1:53

20:38:00 20-05-2021 void netif_debug_log(): 69 - net_if_debug -> net interfaces:
lo0: 127.0.0.1
lo0: ::1
lo0: fe80::1
pdp_ip1: fe80::14bb:55f4:4c6b:2898
pdp_ip1: 2409:4032:6d06:c3d4:9e:2384:da50:a9dc
pdp_ip1: 2409:4032:6d06:c3d4:7084:a35c:477:914e
pdp_ip0: 26.60.29.192
pdp_ip0: fe80::8c2:2fa7:36a4:3ea8
pdp_ip0: 2409:4072:6d00:80aa:805:c8ae:9efc:f302
pdp_ip0: 2409:4072:6d00:80aa:8dd0:98b0:71c:8dd
en0: fe80::ca9:5c79:32e4:bf56
en0: 192.168.18.12
awdl0: fe80::249e:93ff:fe0f:ed79
utun0: fe80::e914:9594:a020:2149
utun1: fe80::1aaa:fa59:469a:f52
ipsec2: fe80::fa95:eaff:fea3:a04c
ipsec2: 2409:4032:6d06:c3d4:9e:2384:da50:a9dc
ipsec3: fe80::fa95:eaff:fea3:a04c
ipsec3: 2409:4032:6d06:c3d4:9e:2384:da50:a9dc
llw0: fe80::249e:93ff:fe0f:ed79

20:38:00 20-05-2021 void netrt_debug_log(): 80 - net_rt_debug -> net routes:
Destination Next Hop Iface Type
0.0.0.0/0 192.168.18.1 en0
0.0.0.0/0 26.60.29.192 pdp_ip0
1.1.1.1/0 192.168.18.1 en0
5.39.68.205/0 192.168.18.1 en0
5.39.85.119/0 192.168.18.1 en0
17.57.12.11/0 192.168.18.1 en0
17.57.145.116/0 26.60.29.192 pdp_ip0
17.57.145.116/0 192.168.18.1 en0
17.57.145.117/0 192.168.18.1 en0
17.110.234.77/0 192.168.18.1 en0
17.242.2.32/0 192.168.18.1 en0
17.242.184.25/0 192.168.18.1 en0
17.248.162.4/0 192.168.18.1 en0
17.248.162.5/0 192.168.18.1 en0
17.248.162.6/0 192.168.18.1 en0
17.248.162.8/0 192.168.18.1 en0
17.248.162.38/0 192.168.18.1 en0
17.248.162.68/0 192.168.18.1 en0
17.248.162.71/0 192.168.18.1

void net_conf_debug(): 47 - net_conf_debug ->
SIP
sip_listen
sip_certificate
sip_cafile
sip_capath
sip_trans_def UDP
sip_verify_server no
sip_tos 160
Call
call_local_timeout 120
call_max_calls 4
call_hold_other_calls yes
Audio
audio_path /share/baresip
audio_player audiounit,nil
audio_source ,
audio_alert audiounit,nil
auplay_srate 48000
ausrc_srate 48000
auplay_channels 0
ausrc_channels 0
audio_level no
Video
video_source ,
#video_source avformat,rtmp://127.0.0.1/app/foo
video_display ,
video_size "640x480"
video_bitrate 1000000
video_fps 30.00
video_fullscreen yes
videnc_format yuv420p
AVT
rtp_tos 184
rtp_video_tos 136
rtp_ports 1024-49152
rtp_bandwidth 0-0
rtcp_mux no
jitter_buffer_type fixed
jitter_buffer_delay 5-10
jitter_buffer_wish 0
rtp_stats no
rtp_timeout 0 # in seconds
Network
net_interface

`

Thanks,
Kannan

Baresip 1.0.0 setup iOS video codec issue

Audio is working fine in the application, now I am trying to implement video. I just follow the steps:
In contrib.mk I just added the following module :

EXTRA_MODULES='g711 audiounit avcapture vp8 vp9'

but getting error one by one ..

modules/avcodec/avcodec.c:9:10: fatal error: 'libavutil/pixdesc.h' file not found #include <libavutil/pixdesc.h

CC [m] /Users/samir/Desktop/WorkFrom_Home_Doc/Baresip1.0.0_Setup/GithubHelp/baresip-ios/build/aarch64/baresip/modules/vp8/decode.o modules/vp8/decode.c:11:10: fatal error: 'vpx/vpx_decoder.h' file not found #include <vpx/vpx_decoder.h>

is there any other module I need to load for mac / iOS? If I don't load the module I will not get video codecs.

unable to compile...

Hi, I'm facing a weird issue...

compilation seems to work fine until

CC /Users/mgs/Project/libp2p/baresip-ios-master/build/armv6/libre/sys/fs.o
CC /Users/mgs/Project/libp2p/baresip-ios-master/build/armv6/libre/sys/rand.o
CC /Users/mgs/Project/libp2p/baresip-ios-master/build/armv6/libre/sys/sleep.o
CC /Users/mgs/Project/libp2p/baresip-ios-master/build/armv6/libre/sys/sys.o
CC /Users/mgs/Project/libp2p/baresip-ios-master/build/armv6/libre/lock/rwlock.o
src/lock/rwlock.c:114: fatal error: opening dependency file /Users/mgs/Project/libp2p/baresip-ios-master/build/armv6/libre/lock/rwlock.d: No such file or directory
compilation terminated.
make[1]: *** [/Users/mgs/Project/libp2p/baresip-ios-master/build/armv6/libre/lock/rwlock.o] Error 1
make: *** [libre] Error 2

I have spent a lot of hours on this... I'm sure it is right on my face...
how come it works fine with all files before rwlock and does not for this one ?

Thank you for your help

Cant use audio from background

Hello, I am trying to connect to call from background, I get VoIP notification and after accepting the call I start app and get following error:
CALL_ANSWERED
stream: update 'audio'
audio: Set audio decoder: PCMU 8000Hz 1ch
�[31maudiounit: AudioSessionSetActive: 1701737535
�[;m�[31maudio: start_player failed (audiounit.default): Operation not supported
�[;m�[31mcall: update: audio_decoder_set error: Operation not supported
�[;maudio: Set audio encoder: PCMU 8000Hz 1ch
�[31maudiounit: AudioSessionSetActive: 1701737535
�[;m�[31maudio: start_player failed (audiounit.default): Operation not supported
�[;m�[31maudiounit: AudioSessionSetActive: 1701737535
�[;m�[31maudio: start_source failed (audiounit.default): Operation not supported
�[;msip:*[email protected]: session closed: Operation not supported
CALL_CLOSED

When I am doing same from foreground it works.

iOS audio problems

Hi all, i'm testing baresip to be the base of a voip service. I based my tests in something similar like this:
https://gist.github.com/ramki1979/6112198

Basically i don't use ua_connect function because i need all the handlers of the call...
Wtih this setup i can establish a call but i can't hear any audio between two real devices, here is the console output:

stdio.so: No such file or directory
aucodec: PCMA/8000/1
aufilt: vumeter
auplay: coreaudio
ausrc: coreaudio

vidsrc: avcapture

�[;m�[31mmodule opengl.so: No such file or directory
�[;mmedianat: stun
medianat: turn
medianat: ice

�[;mPopulated 2 audio codecs
Populated 1 audio filter
Populated 0 video codecs
Populated 0 video filters
registering [email protected]...
register reply: 200 OK
inviting sip:[email protected]...
session progress: 100 Giving a try
session progress: 100 Trying
session progress: 100 Giving a try
SDP answer received
SDP peer address: 81.23.228.129:54272
SDP media format: PCMU/8000/1 (payload type: 0)
session established
register reply: 200 OK

Thanks in advance, any help is welcome! :) @alfredh

Invalid password (0 - 63 characters followed by newline) and outbound requires valid UUID!

When trying to register I am getting an invalid password error

this is my Address:

let addr = "sip:\(username)@\(strSIPURL):18002;auth_pass=\(password);sipnat=outbound;outbound=\"sip:\(strSIPURL):18002;transport=udp\";answermode=manual;stunuser=311;audio_codecs=PCMU/8000/1,PCMA/8000/1;video_codecs=VP9,VP8,H264,H264;mwi=no;ptime=20;regint=3600;regq=0.5;pubint=0;"

My Log:
Local network address: IPv4=:10.14.147.68 IPv6=:2405:204:898b:4f8c:43a:b442:843:8293
�[31mdl: mod: ./stdio.so (dlopen(//stdio.so, 6): image not found)
�[;m�[31mmodule stdio.so: No such file or directory
�[;maucodec: PCMU/8000/1
aucodec: PCMA/8000/1
aufilt: vumeter
auplay: coreaudio
ausrc: coreaudio
vidsrc: avcapture
�[31mdl: mod: ./opengl.so (dlopen(//opengl.so, 6): image not found)
�[;m�[31mmodule opengl.so: No such file or directory
�[;mmedianat: stun
medianat: turn
medianat: ice
�[31muuid: fopen() /uuid (Operation not permitted)
�[;m�[31mdl: mod: ./uuid.so (dlopen(//uuid.so, 6): image not found)
�[;m�[31mmodule uuid.so: No such file or directory
�[;maccount: creating accounts template /accounts
�[31mdl: mod: ./account.so (dlopen(//account.so, 6): image not found)
�[;m�[31mmodule account.so: No such file or directory
�[;mcontact: creating contacts template /contacts
�[31mdl: mod: ./contact.so (dlopen(//contact.so, 6): image not found)
�[;m�[31mmodule contact.so: No such file or directory
�[;mPopulated 2 audio codecs
Populated 1 audio filter
Populated 0 video codecs
Populated 0 video filters
�[31maccount: video codec not found: VP9
�[;m�[31maccount: video codec not found: VP8
�[;m�[31maccount: video codec not found: H264
�[;m�[31maccount: video codec not found: H264

�[;mPlease enter password for [email protected]: Invalid password (0 - 63 characters followed by newline)
`

Then I make the following changes to my address

let addr = "sip:\(username):\(password)@\(strSIPURL):18002;sipnat=outbound;outbound=\"sip:\(strSIPURL):18002;transport=udp\";answermode=manual;stunuser=311;audio_codecs=PCMU/8000/1,PCMA/8000/1;video_codecs=VP9,VP8,H264,H264;mwi=no;ptime=20;regint=3600;regq=0.5;pubint=0;"

and Getting the following error:

Local network address: IPv4=:10.14.147.68 IPv6=:2405:204:898b:4f8c:43a:b442:843:8293
�[31mdl: mod: ./stdio.so (dlopen(//stdio.so, 6): image not found)
�[;m�[31mmodule stdio.so: No such file or directory
�[;maucodec: PCMU/8000/1
aucodec: PCMA/8000/1
aufilt: vumeter
auplay: coreaudio
ausrc: coreaudio
vidsrc: avcapture
�[31mdl: mod: ./opengl.so (dlopen(//opengl.so, 6): image not found)
�[;m�[31mmodule opengl.so: No such file or directory
�[;mmedianat: stun
medianat: turn
medianat: ice
�[31muuid: fopen() /uuid (Operation not permitted)
�[;m�[31mdl: mod: ./uuid.so (dlopen(//uuid.so, 6): image not found)
�[;m�[31mmodule uuid.so: No such file or directory
�[;maccount: creating accounts template /accounts
�[31mdl: mod: ./account.so (dlopen(//account.so, 6): image not found)
�[;m�[31mmodule account.so: No such file or directory
�[;mcontact: creating contacts template /contacts
�[31mdl: mod: ./contact.so (dlopen(//contact.so, 6): image not found)
�[;m�[31mmodule contact.so: No such file or directory
�[;mPopulated 2 audio codecs
Populated 1 audio filter
Populated 0 video codecs
Populated 0 video filters
�[31maccount: video codec not found: VP9
�[;m�[31maccount: video codec not found: VP8
�[;m�[31maccount: video codec not found: H264
�[;m�[31maccount: video codec not found: H264

�[;[email protected]: Using sipnat: `outbound'
�[31mua: outbound requires valid UUID!

OS I am using : MacOS 10.15.2
Xcode Version : 10.3
baresip Version : baresip v0.6.5

31mcall: sipsess_connect: Invalid argument

I have implemented baresip in one of the ios projects, registration is successful for sip account using following code

`

  var agent: OpaquePointer? = nil

Method called -- Client(username: username, password: password, agent: &agent)

 required init?(username: String, password: String, agent: inout OpaquePointer?) throws {
    guard libre_init() == 0 else { throw SipError.libre }
    
    // Initialize dynamic modules.
    mod_init()
    
    // Make configure file.
    if let path = NSSearchPathForDirectoriesInDomains(.documentDirectory, .userDomainMask, true).first {
        conf_path_set(path)
    }
    guard conf_configure() == 0 else { throw SipError.config }
    
    guard baresip_init(conf_config()) == 0 else { print("baresip init error"); return }

    // Initialize the SIP stack.
    guard ua_init("SIP", 1, 1, 1) == 0 else { throw SipError.stack }
    
    // Load modules.
    guard conf_modules() == 0 else { throw SipError.modules }
    
    let addr = "sip:\(username):\(password)@xxxxxx.xxxxx.xxx;transport=udp;answermode=manual"

    // Start user agent.
    guard ua_alloc(&agent, addr) == 0 else { throw SipError.userAgent }
    
    uag_event_register({ (userAgent, event, call, prm, arg) in
        print(event)
        if event.rawValue == 6 {
            DispatchQueue.main.async {
                if let handler = SipClient.incommingCallHandler {
                    handler(call)
                }
            }
        }
        if event.rawValue == 10 {
            print("Call Closed")
            DispatchQueue.main.async {
                if let handler = SipClient.callEnded {
                    //re_cancel()
                    handler()
                }
            }
        }
    }, nil)
    
    let registered = ua_isregistered(agent)
    if registered == 0 {
        print("USER \(username)@xxx.xxxx.xxxx REGISTERED!!!!!")
    } else {
        print("USER REGISTERATION FAILED!!!")
    }
    
    DispatchQueue.global(qos: .userInitiated).async {
        re_main(nil)
    }
}

`

now I am trying to make a call and I wrote following function for the same.
`

 func makeCall(agent: inout OpaquePointer?, toUri: String) {

        print("sip:\(toUri)@xxxxx.xxxx.xxxx");

            guard ua_connect(agent, nil, nil, "sip:\(toUri)@xxxxx.xxxxx.xxx", VIDMODE_OFF) == 0 else { return }
        
            call_set_handlers(ua_call(uag_current()), { (call, call_event, str, arg) in
                print("Call event: ")
                print(call_event)
                if call_event.rawValue == 3 {
                    print("Call Established")
                    DispatchQueue.main.asyncAfter(deadline: .now() + 8, execute: {
                        let duration = call_duration(call)
                        print("call duration: \(duration)")
                    })
                }
                
                if call_event.rawValue == 4 {
                    if let handler = SipClient.callEnded {
                        ua_hangup(call_get_ua(call), call, 0, "Call Disconnected By peer")
                        handler()
                    }
                }
            }, { (call, key, arg) in
                //            print(call)
                //            print(key)
                //            print(arg)
            }, nil)
        }

`
but when the method ua_connect called, it gives following error
call: connecting to 'sip:[email protected]'..

�[31mcall: sipsess_connect: Invalid argument
Could you please help me for the above issue

@alfredh @iwanbk @Anakros @sreimers

Does video work for Android ?

It seems that Camera and Video display modules do not support Android yet. Is that right ? Or I am missing something.

Build libraries for openSSL/TLS

I try to use baresip in an iOS app, but need to connect with TLS. However, the lib's build with this makefile are excluding openSSL and TLS. Why is that, and how can I build lib's that use TLS?

RE: dyld: Library not loaded: libre.dylib Reason: image not found

I have the same issue compile IOS App just as the one here.
I have followed his steps(or solution) and I have this as my error

dyld: Library not loaded: libre.dylib Referenced from: /Users/<apple>/Library/Developer/CoreSimulator/Devices/19EE7E1B-1236-4E74-A998-9F29186286C9/data/Containers/Bundle/Application/B366E9C2-51ED-4B2F-ACCE-6E981EACDE82/baresip-Test.app/baresip-Test Reason: image not found.

Anyone here know how to solve this problem.

OSX: 10.15.5
XCode: Version 11.5 (11E608c)
Baresip: Latest

dyld: Library not loaded: libre.dylib Reason: image not found

I followed your guidelines and was able to make library.

I draged the resulted armv7 folder to my project and added single line in AppDelegate's didFinishLaunchingWithOptions method..

/* Initialise System library */
int err = libre_init();

when I command + click on this method, it reveals the method in ...armv7/include/re/re_main.h file.

I'm able to compile it. but when I run it crashes and shows following debug info in logs

dyld: Library not loaded: libre.dylib
Referenced from: /var/mobile/Applications/C43EE39E-9DAA-4F4D-9EF9-4A41A8A5BAED/my.app/my
Reason: image not found

Here is screenshot of my project's Linked Framework and Libraries section.:

Screen.png

Any insights to this problem will be much appreciated.

As I found this which said to make library optional didn't work out as when execution reaches on the above line of code, app crashes again.

Unexpected timeouts on canceled connecting call

Hello. I am using baresip v 1.0.0 as a library (on iOS so with kqueue). My use cases require to stop baresip main loop and restart when required. I have found the edge case when on finishing the call which is trying to connect and trying to shut down baresip completely I have some delay which may take about up to 20 seconds or so. So the next call I can start only after the main loop finished and I start the new one. The sequence is something like this:

  • start baresip main loop
  • ua_connect
  • the call is trying to connect
  • ua_hangup
  • ua_stop_all(0) inside mqueue callback
  • here I get a long delay until re_main returns

I do not get a delay in case the call was connected and then finished using ua_hangup. So probably there are some issues only when a call can not be connected and start waiting for something to happen.

Are there any timeout settings or so?

Enable Speaker on iPhone

Hi,

I have worked on Baresip for some days to build a Sip client on iPhone. It's working great!
Now, I want to implement a button that be able to Mute or using Speaker, how can do it?

Thanks,
Tuan.

Compile error: re.h and rem_fir.h missing

The following error is observed during the build:

CC /Users/ttarhan/Development/OpenSource/baresip/build/armv7/librem/fir/fir.o
src/fir/fir.c:9:16: error: re.h: No such file or directory
src/fir/fir.c:10:21: error: rem_fir.h: No such file or directory

No such file or directory in conf_configure()

I am creating an application for free audio and video calling using baresip, I have created demo and its working fine in Swift 4.2 , Now same thing I am applying in my existing project and its give me strange behaviour, existing project is in Swift 3.0, below is my code.

 `   override func viewDidLoad() {
      super.viewDidLoad()

    guard libre_init() == 0 else { return }
    
    // Initialize dynamic modules
    mod_init()
    
    // make configure file
    
    if let path = NSSearchPathForDirectoriesInDomains(.documentDirectory, .userDomainMask, true).first {
        conf_path_set(path)
    }
    
    guard conf_configure() == 0 else { return }
    
    guard baresip_init(conf_config(), 0) == 0 else { return }
    
    //Initialize the SIP Account
    
    guard ua_init("SIP",1,1,1,0) == 0 else { return }
    
    // Load Modules
    
    guard conf_modules() == 0 else { return }
    
    let addr = "sip:[email protected];auth_pass=IB5zbrzi;transport=upd;answermode=manual;audio_codecs=PCMU/8000/1;audio_codecs=PCMA/8000/1"
    
    // Start user agent.
    guard ua_alloc(&agent, addr) == 0 else { return }
    
    let registered = ua_isregistered(agent)
    if registered == 0 {
        print("USER [email protected] REGISTERED!!!!!")
    } else {
        print("USER REGISTERATION FAILED!!!")
    }
    
    uag_event_register({ (agent, event, call, param, arg) in
        print("User Agent \(event)")
    }, nil)
    
    call_set_handlers(ua_call(agent), { (call, call_event, str, arg) in
        print("Call event: ")
        print(call_event)
        
        if call_event.rawValue == 0 {
            print("Incomming call from \(str): \(arg)")
        }
        
        if call_event.rawValue == 4 {
            //                re_cancel()
            ua_hangup(call_get_ua(call), call, 0, "Call Disconnected By peer")
        }
    }, { (call, key, arg) in
        //            print(call)
        //            print(key)
        //            print(arg)
    }, nil)
    
    // Start the main loop.
    DispatchQueue.global(qos: .userInitiated).async {
        re_main(nil)
    }
}`

 `///////Out put in NSLog

  Printing description of path: "/var/mobile/Containers/Data/Application/904E1C0B-EEC4.    -4642-B61A-079A9FBC5D92/Documents" 
   config: creating config template UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUA\247cء!/config 

  [31mconfig: writing UUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUUA\247cء!/config: No such file or      directory`

meke contrib: error: missing MD5 backend

libre/dns/darwin/srv.o
CC /Users/zhangxinyao/Documents/project/baresip/baresip-ios/build/aarch64/libre/md5/wrap.o
src/md5/wrap.c:39:2: error: missing MD5 backend
#error missing MD5 backend
^
src/md5/wrap.c:25:25: warning: unused parameter 'd' [-Wunused-parameter]
void md5(const uint8_t *d, size_t n, uint8_t *md)
^
src/md5/wrap.c:25:35: warning: unused parameter 'n' [-Wunused-parameter]
void md5(const uint8_t *d, size_t n, uint8_t *md)
^
src/md5/wrap.c:25:47: warning: unused parameter 'md' [-Wunused-parameter]
void md5(const uint8_t *d, size_t n, uint8_t *md)
^
3 warnings and 1 error generated.
make[1]: *** [/Users/uname/Documents/project/baresip/baresip-ios/build/aarch64/libre/md5/wrap.o] Error 1
make: *** [libre] Error 2

Memory leaks and crash after relaunching the baresip stack in iOS

If I register or make a call and then close baresip and relaunch it again it will crash after some time has passed or if I try to register or make a call again.
My code is here: https://pastebin.com/t7a80DFz
and the log below

31mconfig: no sip_cafile defined, tls connections maybe won't work
�[;mLocal network address:  IPv4=en0|192.168.1.7 
�[31mdl: mod: ./stdio.so (dlopen(//stdio.so, 6): image not found)
�[;m�[31mmodule stdio.so: No such file or directory
�[;maucodec: PCMU/8000/1
aucodec: PCMA/8000/1
auplay: audiounit
ausrc: audiounit
medianat: stun
medianat: turn
medianat: ice
�[31muuid: fopen() per/CoreSimulator/Devices/3AA0D104-52FD-4790-8A14-F439277D1DA5/data/Containers/Bundle/Application/5BA743CA-0A66-4E19-B5F9-EF4A254C9495/baresip2nattempt.app/Frameworks/uuid (No such file or directory)
�[;m�[31mdl: mod: ./uuid.so (dlopen(//uuid.so, 6): image not found)
�[;m�[31mmodule uuid.so: No such file or directory
�[;maccount: creating accounts template per/CoreSimulator/Devices/3AA0D104-52FD-4790-8A14-F439277D1DA5/data/Containers/Bundle/Application/5BA743CA-0A66-4E19-B5F9-EF4A254C9495/baresip2nattempt.app/Frameworks/accounts
�[31mdl: mod: ./account.so (dlopen(//account.so, 6): image not found)
�[;m�[31mmodule account.so: No such file or directory
�[;mcontact: creating contacts template per/CoreSimulator/Devices/3AA0D104-52FD-4790-8A14-F439277D1DA5/data/Containers/Bundle/Application/5BA743CA-0A66-4E19-B5F9-EF4A254C9495/baresip2nattempt.app/Frameworks/contacts
�[31mdl: mod: ./contact.so (dlopen(//contact.so, 6): image not found)
�[;m�[31mmodule contact.so: No such file or directory
�[;mPopulated 2 audio codecs
Populated 0 audio filters
Populated 0 video codecs
Populated 0 video filters
[email protected]: (prio 0) {0/UDP/v4} 200 OK (Grandstream UCM6202V1.7A 1.0.20.23) [1 binding]
"error value 0"
ua: stop all (forced=0)
"uag exit handler"
fd 6 in use: flags=1 fh=0x10a014810 arg=0x6000014c8dc0
fd 7 in use: flags=1 fh=0x10a014820 arg=0x6000014c9870
fd 8 in use: flags=1 fh=0x10a014810 arg=0x6000014c4dc0
fd 9 in use: flags=1 fh=0x10a016cf0 arg=0x6000002c36a0
Timers (2):
  0x7fb13c618578: th=0x109fe5480 expire=500ms
  0x7fb13c618540: th=0x109fe5440 expire=32000ms
�[31mmem: Memory leaks (60):
�[;m  0x7fb13d059e40: nrefs=1  size=1080    [10 00 00 00 02 00 00 00 10 27 00 00 30 75 00 00 ] [.........'..0u..]
  0x6000014c8dc0: nrefs=1  size=72      [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000014c9870: nrefs=1  size=72      [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000005c6840: nrefs=1  size=16      [90 99 72 3c b1 7f 00 00 10 00 00 00 00 00 00 00 ] [..r<............]
  0x7fb13c729990: nrefs=1  size=256     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000005c7060: nrefs=1  size=16      [e0 e9 2d 00 00 60 00 00 02 00 00 00 00 00 00 00 ] [..-..`..........]
  0x6000002de9e0: nrefs=1  size=32      [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x600001ff8040: nrefs=1  size=152     [e0 75 1c 01 00 60 00 00 80 74 1c 01 00 60 00 00 ] [.u...`...t...`..]
  0x6000005ed670: nrefs=1  size=16      [30 5e 40 3c b1 7f 00 00 10 00 00 00 00 00 00 00 ] [0^@<............]
  0x7fb13c405e30: nrefs=1  size=256     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000008d8140: nrefs=1  size=64      [00 00 00 00 00 00 00 00 c0 81 8d 00 00 60 00 00 ] [.............`..]
  0x6000005eda30: nrefs=1  size=16      [00 47 40 3c b1 7f 00 00 10 00 00 00 00 00 00 00 ] [.G@<............]
  0x7fb13c404700: nrefs=1  size=256     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000008d81c0: nrefs=1  size=64      [40 81 8d 00 00 60 00 00 00 00 00 00 00 00 00 00 ] [@....`..........]
  0x6000005ef150: nrefs=1  size=16      [40 e5 40 3c b1 7f 00 00 10 00 00 00 00 00 00 00 ] [@.@<............]
  0x7fb13c40e540: nrefs=1  size=256     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000005ef1a0: nrefs=1  size=16      [60 5e 41 3c b1 7f 00 00 10 00 00 00 00 00 00 00 ] [`^A<............]
  0x7fb13c415e60: nrefs=1  size=256     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000005ecf40: nrefs=1  size=16      [30 c6 40 3c b1 7f 00 00 10 00 00 00 00 00 00 00 ] [0.@<............]
  0x7fb13c40c630: nrefs=1  size=256     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000008d8240: nrefs=1  size=56      [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000005ef240: nrefs=1  size=4       [53 49 50 00                                     ] [SIP.            ]
  0x6000002c30a0: nrefs=1  size=24      [50 aa fe 09 01 00 00 00 40 80 ff 01 00 60 00 00 ] [P.......@....`..]
  0x6000011c75e0: nrefs=1  size=112     [00 00 00 00 00 00 00 00 30 75 1c 01 00 60 00 00 ] [........0u...`..]
  0x6000014c4dc0: nrefs=1  size=72      [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000011c7530: nrefs=1  size=112     [e0 75 1c 01 00 60 00 00 80 74 1c 01 00 60 00 00 ] [.u...`...t...`..]
  0x6000002c36a0: nrefs=1  size=24      [09 00 00 00 ff ff ff ff d0 b9 fe 09 01 00 00 00 ] [................]
  0x6000011c7480: nrefs=1  size=112     [30 75 1c 01 00 60 00 00 00 00 00 00 00 00 00 00 ] [0u...`..........]
  0x7fb13d80f440: nrefs=1  size=1176    [90 07 4c 01 00 60 00 00 00 00 00 00 00 00 00 00 ] [..L..`..........]
  0x6000014c0790: nrefs=1  size=65      [73 69 70 3a 31 30 30 31 3a 4e 4f 56 31 30 30 31 ] [sip:1001:NOV1001]
  0x600000dc2030: nrefs=1  size=39      [73 69 70 3a 31 30 30 31 40 6e 6f 76 69 74 61 65 ] [sip:1001@novitae]
  0x6000005c1350: nrefs=2  size=8       [4e 4f 56 31 30 30 31 00                         ] [NOV1001.        ]
  0x7fb13c406be0: nrefs=1  size=216     [00 00 00 00 00 00 00 00 10 02 de 3f c0 a8 01 07 ] [...........?....]
  0x600001ac4b40: nrefs=1  size=184     [10 76 dc 00 00 60 00 00 03 00 00 00 00 00 00 00 ] [.v...`..........]
  0x600000dc7610: nrefs=1  size=33      [73 69 70 3a 6e 6f 76 69 74 61 65 32 2e 6d 79 63 ] [sip:novitae2.myc]
  0x6000002c2a40: nrefs=1  size=17      [35 62 33 65 31 65 66 35 34 62 36 32 39 35 39 66 ] [5b3e1ef54b62959f]
  0x6000002c32e0: nrefs=1  size=17      [32 32 39 66 32 34 64 66 37 65 32 64 62 65 33 61 ] [229f24df7e2dbe3a]
  0x6000002c2e60: nrefs=1  size=32      [60 62 41 3c b1 7f 00 00 00 02 00 00 00 00 00 00 ] [`bA<............]
  0x7fb13c416260: nrefs=1  size=512     [54 6f 3a 20 3c 73 69 70 3a 31 30 30 31 40 6e 6f ] [To: <sip:1001@no]
  0x600000dc7450: nrefs=1  size=40      [20 6f 7c 01 00 60 00 00 20 6f 7c 01 00 60 00 00 ] [ o|..`.. o|..`..]
  0x6000002c2560: nrefs=1  size=20      [31 30 30 31 2d 30 78 37 66 62 31 33 63 35 30 61 ] [1001-0x7fb13c50a]
  0x6000008d8640: nrefs=1  size=64      [2b 73 69 70 2e 69 6e 73 74 61 6e 63 65 3d 22 3c ] [+sip.instance="<]
  0x6000002c2e00: nrefs=1  size=32      [60 b4 41 3c b1 7f 00 00 00 01 00 00 00 00 00 00 ] [`.A<............]
  0x7fb13c41b460: nrefs=1  size=256     [41 6c 6c 6f 77 3a 20 49 4e 56 49 54 45 2c 41 43 ] [Allow: INVITE,AC]
  0x6000017c6f20: nrefs=1  size=88      [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000005ed120: nrefs=1  size=12      [67 72 61 6e 64 73 74 72 65 61 6d 00             ] [grandstream.    ]
  0x6000005ef880: nrefs=1  size=5       [31 30 30 31 00                                  ] [1001.           ]
  0x600000dc7760: nrefs=1  size=44      [31 36 31 33 39 34 36 39 32 30 2f 38 66 37 39 37 ] [1613946920/8f797]
  0x6000005ee7f0: nrefs=1  size=5       [61 75 74 68 00                                  ] [auth.           ]
  0x6000002c3460: nrefs=1  size=17      [37 33 66 32 30 66 32 64 30 35 62 39 32 65 37 37 ] [73f20f2d05b92e77]
  0x6000002ccee0: nrefs=1  size=32      [40 4a 82 3d b1 7f 00 00 00 08 00 00 00 00 00 00 ] [@J.=............]
  0x7fb13d824a40: nrefs=1  size=2048    [4d 61 78 2d 46 6f 72 77 61 72 64 73 3a 20 37 30 ] [Max-Forwards: 70]
  0x7fb13c508cd0: nrefs=1  size=200     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
  0x6000005c1df0: nrefs=2  size=9       [52 45 47 49 53 54 45 52 00                      ] [REGISTER.       ]
  0x600000dc1cb0: nrefs=1  size=33      [73 69 70 3a 6e 6f 76 69 74 61 65 32 2e 6d 79 63 ] [sip:novitae2.myc]
  0x6000002cc940: nrefs=2  size=24      [6e 6f 76 69 74 61 65 32 2e 6d 79 63 72 65 73 74 ] [novitae2.mycrest]
  0x6000002c9b40: nrefs=1  size=24      [7a 39 68 47 34 62 4b 35 34 33 66 36 35 31 33 30 ] [z9hG4bK543f65130]
  0x6000002c9f00: nrefs=1  size=32      [40 c0 03 3e b1 7f 00 00 00 04 00 00 00 00 00 00 ] [@..>............]
  0x7fb13e03c040: nrefs=1  size=1024    [52 45 47 49 53 54 45 52 20 73 69 70 3a 6e 6f 76 ] [REGISTER sip:nov]
  0x7fb13c618500: nrefs=1  size=272     [00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ] [................]
"deinit sipclient"
"sip state has been deinited"
�[31mconfig: no sip_cafile defined, tls connections maybe won't work
�[;mLocal network address:  IPv4=en0|192.168.1.7 
�[31mdl: mod: ./stdio.so (dlopen(//stdio.so, 6): image not found)
�[;m�[31mmodule stdio.so: No such file or directory
�[;maucodec: PCMU/8000/1
aucodec: PCMA/8000/1
auplay: audiounit
ausrc: audiounit
medianat: stun
medianat: turn
medianat: ice
�[31muuid: fopen() per/CoreSimulator/Devices/3AA0D104-52FD-4790-8A14-F439277D1DA5/data/Containers/Bundle/Application/5BA743CA-0A66-4E19-B5F9-EF4A254C9495/baresip2nattempt.app/Frameworks/uuid (No such file or directory)
�[;m�[31mdl: mod: ./uuid.so (dlopen(//uuid.so, 6): image not found)
�[;m�[31mmodule uuid.so: No such file or directory
�[;maccount: creating accounts template per/CoreSimulator/Devices/3AA0D104-52FD-4790-8A14-F439277D1DA5/data/Containers/Bundle/Application/5BA743CA-0A66-4E19-B5F9-EF4A254C9495/baresip2nattempt.app/Frameworks/accounts
�[31mdl: mod: ./account.so (dlopen(//account.so, 6): image not found)
�[;m�[31mmodule account.so: No such file or directory
�[;mcontact: creating contacts template per/CoreSimulator/Devices/3AA0D104-52FD-4790-8A14-F439277D1DA5/data/Containers/Bundle/Application/5BA743CA-0A66-4E19-B5F9-EF4A254C9495/baresip2nattempt.app/Frameworks/contacts
�[31mdl: mod: ./contact.so (dlopen(//contact.so, 6): image not found)
�[;m�[31mmodule contact.so: No such file or directory
�[;mPopulated 2 audio codecs
Populated 0 audio filters
Populated 0 video codecs
Populated 0 video filters
�[31mmem: mem_deref: magic check failed 0xb5b5b5b5 (0x7fb13c618500)
baresip2nattempt was compiled with optimization - stepping may behave oddly; variables may not be available.
�[;m(lldb) thread backtrace
* thread baresip/baresip#15, queue = 'com.apple.root.user-initiated-qos', stop reason = EXC_BREAKPOINT (code=EXC_I386_BPT, subcode=0x0)
  * frame #0: 0x000000010a0205d3 baresip2nattempt`mem_deref(data=0x00007fb13c618500) at mem.c:314:6 [opt]
    frame baresip/baresip#1: 0x000000010a01e6d6 baresip2nattempt`tmr_poll(tmrl=0x000000010a04b200) at tmr.c:115:3 [opt]
    frame baresip/baresip#2: 0x000000010a01fa70 baresip2nattempt`re_main(signalh=<unavailable>) at main.c:1040:3 [opt]
    frame baresip/baresip#3: 0x0000000109fac919 baresip2nattempt`closure baresip/baresip#2 in SipClient.init(self=0x00006000033901c0) at ViewController.swift:103:30
    frame baresip/baresip#4: 0x0000000109facbe0 baresip2nattempt`thunk for @escaping @callee_guaranteed () -> () at <compiler-generated>:0
    frame baresip/baresip#5: 0x000000010a32b7ec libdispatch.dylib`_dispatch_call_block_and_release + 12
    frame baresip/baresip#6: 0x000000010a32c9c8 libdispatch.dylib`_dispatch_client_callout + 8
    frame baresip/baresip#7: 0x000000010a33e6e4 libdispatch.dylib`_dispatch_root_queue_drain + 827
    frame baresip/baresip#8: 0x000000010a33ee6d libdispatch.dylib`_dispatch_worker_thread2 + 135
    frame baresip/baresip#9: 0x00007fff60c8f47a libsystem_pthread.dylib`_pthread_wqthread + 244
    frame baresip/baresip#10: 0x00007fff60c8e493 libsystem_pthread.dylib`start_wqthread + 15
(lldb)

re.h: No such file or directory.

When I am using $ sudo make install command in Ubuntu, I am getting this error
fatal error: re.h: No such file or directory.
I am using android studio

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