Problem with VideoCall

Aing Tii

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May 22, 2021, 4:21:17 AM5/22/21

to baresip

If I start video Call and answer on my phone, it will emidiatly hangUp.

protokoll:

All 1 useragent registered successfully! (189 ms)

/dialdir 100 audio=sendrecv video=sendonly

1...@192.168.5.82: selected for request

ua: using best effort AF: af=AF_INET

call: alloc with params laddr=192.168.5.79, af=AF_INET, use_rtp=1

call: use_video=1

call: connecting to 'sip:1...@192.168.5.82:5062'..

call id: a2f27939b21dc7cf

call: SIP Progress: 100 Trying (/)

call: SIP Progress: 180 Ringing (/)

video: stopping video source ..

video: stopping video display ..

alsa: reset: srate=8000, ch=1, num_frames=320, pcmfmt=S16_LE

alsa: playback started (default)

stream: incoming rtp for 'audio' established, receiving from 192.168.5.82:7380

audio: Set audio decoder: PCMU 8000Hz 1ch

audio: start

audio: enable auplay resampler: 8000Hz/1ch --> 48000Hz/2ch

audio: create recv buffer [20 - 160 ms] [3840 - 30720 bytes]

alsa: reset: srate=48000, ch=2, num_frames=960, pcmfmt=S16_LE

alsa: playback started (default)

audio: player started with sample format S16LE

call: got SDP answer (418 bytes)

1...@192.168.5.82: Call answered: sip:1...@192.168.5.82:5062

alsa: stopping playback thread (default)

call: update media

stream: update 'audio'

stream: audio: starting RTCP with remote 192.168.5.82:7381

stream: update 'video'

stream: video: starting RTCP with remote 192.168.5.82:7383

audio: update

audio: Set audio encoder: PCMU 8000Hz 1ch

audio: start

audio: enable ausrc resampler: 8000Hz/1ch <-- 48000Hz/2ch

alsa: reset: srate=48000, ch=2, num_frames=960, pcmfmt=S16_LE

alsa: recording started (default) format=S16LE

audio: source started with sample format S16LE

audio tx pipeline: alsa ---> vumeter ---> PCMU

audio rx pipeline: alsa <--- vumeter <--- PCMU

video: update

Set video encoder: VP8 (500000 bit/s, 30.00 fps)

video: start source

v4l2: /dev/video0: found valid V4L2 device (640 x 480) pixfmt=YU12

v4l2: Current input: "Camera 0"

v4l2: current framerate is 30.00 fps

video: stopping video display ..

call: stream start (active=1)

audio: start

video: update

video: stopping video display ..

1...@192.168.5.82: Call established: sip:1...@192.168.5.82:5062

call: reinvite: could not decode SDP offer: Operation not supported== ] [== ]

sip:1...@192.168.5.82:5062: session closed: Connection reset by peer

alsa: stopping recording thread (default)

alsa: stopping playback thread (default)

video: stopping video source ..

v4l2: stopping video source..

video: stopping video display ..

sip:1...@192.168.5.82:5062: Call with sip:1...@192.168.5.82:5062 terminated (duration: 32 secs)

video: stopping video source ..

video: stopping video display ..

audio: destroyed (started=0)

config:

#

# baresip configuration

#

#------------------------------------------------------------------------------

# Core

poll_method epoll # poll, select, epoll ..

# SIP

sip_trans_bsize 128

#sip_listen 0.0.0.0:5060

#sip_certificate cert.pem

# Call

call_local_timeout 120

call_max_calls 4

# Audio

#audio_path /usr/share/baresip

audio_player alsa,default

audio_source alsa,default

audio_alert alsa,default

ausrc_srate 48000

auplay_srate 48000

ausrc_channels 2

auplay_channels 2

#audio_txmode poll # poll, thread

audio_level no

ausrc_format s16 # s16, float, ..

auplay_format s16 # s16, float, ..

auenc_format s16 # s16, float, ..

audec_format s16 # s16, float, ..

# Video

#video_source v4l2,/dev/video0

#video_display x11,nil

video_size 640x480

video_bitrate 500000

video_fps 30.00

video_fullscreen yes

videnc_format yuv420p

# AVT - Audio/Video Transport

rtp_tos 184

rtp_video_tos 136

#rtp_ports 10000-20000

#rtp_bandwidth 512-1024 # [kbit/s]

rtcp_mux no

jitter_buffer_type fixed # off, fixed, adaptive

jitter_buffer_delay 5-10 # frames

rtp_stats no

#rtp_timeout 60

# Network

#dns_server 10.0.0.1:53

#net_interface eth0

# BFCP

#bfcp_proto udp

#------------------------------------------------------------------------------

# Modules

module_path /usr/lib/baresip/modules

# UI Modules

module stdio.so

#module cons.so

#module evdev.so

#module httpd.so

# Audio codec Modules (in order)

#module opus.so

#module amr.so

#module g7221.so

#module g722.so

#module g726.so

module g711.so

#module gsm.so

#module l16.so

#module bv32.so

#module mpa.so

#module codec2.so

#module ilbc.so

#module isac.so

# Audio filter Modules (in encoding order)

module vumeter.so

#module sndfile.so

#module speex_aec.so

#module speex_pp.so

#module plc.so

# Audio driver Modules

module alsa.so

#module pulse.so

#module jack.so

#module portaudio.so

#module aubridge.so

#module aufile.so

# Video codec Modules (in order)

#module avcodec.so

module vp8.so

#module vp9.so

#module h265.so

# Video filter Modules (in encoding order)

#module selfview.so

#module snapshot.so

#module swscale.so

module vidinfo.so

# Video source modules

#module v4l.so

module v4l2.so

module v4l2_codec.so

#module avformat.so

#module x11grab.so

#module cairo.so

#module vidbridge.so

# Video display modules

#module directfb.so

#module x11.so

#module sdl2.so

#module fakevideo.so

# Audio/Video source modules

#module rst.so

#module gst1.so

#module gst_video1.so

# Media NAT modules

module stun.so

module turn.so

module ice.so

#module natpmp.so

#module pcp.so

# Media encryption modules

#module srtp.so

#module dtls_srtp.so

#module zrtp.so

#------------------------------------------------------------------------------

# Temporary Modules (loaded then unloaded)

module_tmp uuid.so

module_tmp account.so

#------------------------------------------------------------------------------

# Application Modules

module_app auloop.so

#module_app b2bua.so

module_app contact.so

module_app debug_cmd.so

#module_app dtmfio.so

#module_app echo.so

#module_app gtk.so

module_app menu.so

#module_app mwi.so

#module_app natbd.so

#module_app presence.so

#module_app syslog.so

#module_app mqtt.so

module_app ctrl_tcp.so

module_app vidloop.so

#------------------------------------------------------------------------------

# Module parameters

cons_listen 0.0.0.0:5555

http_listen 0.0.0.0:8000

ctrl_tcp_listen 0.0.0.0:4444

evdev_device /dev/input/event0

# Opus codec parameters

opus_bitrate 28000 # 6000-510000

#opus_stereo yes

#opus_sprop_stereo yes

#opus_cbr no

#opus_inband_fec no

#opus_dtx no

#opus_mirror no

#opus_complexity 10

#opus_application audio # {voip,audio}

vumeter_stderr yes

# Selfview

video_selfview window # {window,pip}

selfview_size 64x64

# ICE

ice_turn no

ice_debug no

ice_nomination regular # {regular,aggressive}

ice_mode full # {full,lite}

# avcodec

avcodec_h264enc libx264

avcodec_h264dec h264

#avcodec_h265enc libx265

#avcodec_h265dec hevc

#avcodec_hwaccel vaapi

#avcodec_profile_level_id 42002a

# ZRTP

#zrtp_hash no # Disable SDP zrtp-hash (not recommended)

# Menu

#menu_bell yes

#redial_attempts 3 # Num or <inf>

#redial_delay 5 # Delay in seconds

#ringback_disabled yes

#statmode_default off

Aing Tii

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May 22, 2021, 4:22:38 AM5/22/21

to baresip

accounts

<sip:1...@192.168.5.82:5062>;auth_user=xyz;auth_pass=xyz;video_codecs=vp8

Juha Heinanen

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May 22, 2021, 4:37:13 AM5/22/21

to Aing Tii, baresip

Looks like you have commented video source and display and don't have
any video display modules.

-- Juha

Aing Tii

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May 22, 2021, 4:55:57 AM5/22/21

to baresip

ok, I will look, Do you have suggetion for video display module?

in the log this line ist red:

call: reinvite: could not decode SDP offer: Operation not supported

thank you

Jens

Juha Heinanen

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May 22, 2021, 5:01:53 AM5/22/21

to Aing Tii, baresip

Aing Tii writes:

> ok, I will look, Do you have suggetion for video display module?

x11 if you are on linux.

> in the log this line ist red:
> call: reinvite: could not decode SDP offer: Operation not supported

don't know about that. solve first your other problems.

-- Juha

Aing Tii

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May 24, 2021, 3:22:28 AM5/24/21

to baresip

OK, got it, it possible to make videocall.

it seems I always have to use:
export DISPLAY=:0

and/or ?

set | grep DISPLAY

But...

only if I use this command:
/dialdir 100 audio=sendrecv video=sendrecv

if I use this:
/dialdir 100 audio=sendrecv video=sendonly

its hangup

Is it possible to set that as default, everytime I just call the number?
example: dial 100

Because, if I use NodeRed by tcp-ctr
its not working with the command
/dialdir 100 audio=sendrecv video=sendrecv

maybe I made also somthing wrong in config....?

video_display x11,nil

video_size 320x240

module x11.so

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