No early video after update to main

[frogblueDFi]

Hi!

I recently updated from baresip version 3.0 to 3.2.
Everything fine but I observed that early video is not functional any more. After answering the call, video is ok.
I checked change "menu limit early video #2514" but can't see what I have to change to get it back working.

Any suggestions? Must I add/change configurations? Additional calls to accept early video?

Regards
Dietmar

[sreimers]

Can you test with git main?

This commit may be related: 8347ada

[frogblueDFi]

Thanx for the quick response!

Sorry, I was wrong, I updated to git main, not 3.2. No early video with git main.
Tried now with git tag v3.2.0, early video is functional!!

Dietmar

[sreimers]

CC: @cspiel1 @maximilianfridrich

[alfredh]

https://github.com/baresip/baresip/wiki/Guidelines-for-reporting-Issues

please include config and log

[cspiel1]

yes, we need config, accounts and log. And the steps how this can be reproduced.

[frogblueDFi]

Sorry for not providing config etc.
I use baresip as a library.
Again, v3.2.0 is perfect, main has no early video. Same compilation steps. But only iOS and MacOS. Android not yet available.
Can't provide logs at the moment as I don't have the time today to switch back to main.

Config

SIP

sip_listen
sip_certificate
sip_cafile
sip_capath
sip_transports UDP,TCP,TLS,WS,WSS
sip_trans_def UDP
sip_verify_server no
sip_tos 160

Call

call_local_timeout 120
call_max_calls 1
call_hold_other_calls yes

Audio

audio_path
audio_player coreaudio,
audio_source coreaudio,
audio_alert coreaudio,
auplay_srate 16000
ausrc_srate 16000
auplay_channels 1
ausrc_channels 1
audio_txmode poll
audio_level no
ausrc_format S16LE
auplay_format S16LE
auenc_format S16LE
audec_format S16LE
audio_buffer 20-160 # ms
audio_buffer_mode fixed # fixed, adaptive
audio_silence -35.0 # in [dB]
audio_telev_pt 101

Video

video_source avcapture,Logitech Webcam C925e
#video_source avformat,rtmp://127.0.0.1/app/foo
video_display frogvideo,
video_size "320x240"
video_bitrate 1000000
video_fps 12.00
video_fullscreen no
videnc_format nv12

AVT

rtp_tos 184
rtp_video_tos 136
rtp_ports 1024-49152
rtp_bandwidth 512-4096
audio_jitter_buffer_type fixed
audio_jitter_buffer_delay 5-10
video_jitter_buffer_type fixed
video_jitter_buffer_delay 5-50
rtp_stats yes
rtp_timeout 0 # in seconds

Network

net_interface
net_af unspecified

Account

"diFiMac sip:12-dietmar@sip.dev.frogblue.cloud;transport=tls;auth_pass=regint=60;prio=0;medianat:stun;stunserver=stun:stun.dev.frogblue.cloud:3478"

[cspiel1]

Which answermode did you set? In the above I could not see that early media is activated. This has to be done in the accounts file at the callee (the UA that receives the call).

[frogblueDFi]

We use baresip as lib and set answer mode directly after creating the account.
account_set_answermode(acc, ANSWERMODE_EARLY_VIDEO);
account_set_sip_autoanswer(acc, true);
It's not part of the registration string, but will try it.

[cspiel1]

The lib API is also an option. No need to try the same with accounts file.

You have to choose between answermode early video and autoanswer.

• The first leads to answer with 183 Session Progress and enables the early video before the call is established.
• The later establishes the incoming call immediately with a 200 Ok.

Please close the issue if this answers your questions!

[frogblueDFi]

Hi there!

I would like to "reopen" this issue after a new try to update baresip to v3.8.1 for other reasons.
Still have no early video on the receiving side.
As I was wondering not to receive a UA_EVENT_CALL_REMOTE_SDP ua event on an incoming call with early video I started debugging and comparing with code from version v3.2.0.

The obvious problem (for me) is a change in file call.c function call_progress_dir, line 1296.
Previous code
if (call->got_offer) err = update_media(call);
was changed to
if (call->got_offer) { ua_event(call->ua, UA_EVENT_CALL_LOCAL_SDP, call, "answer"); err = call_update_media(call); }
So previous (functional code) sends the UA_EVENT_CALL_REMOTE_SDP not UA_EVENT_CALL_LOCAL_SDP.
Reverting the change did the fix for me as I rely on the event to indicate if the remote sends video or not.

Regards
Dietmar

[cspiel1]

Please open a new issue and follow the guide lines https://github.com/baresip/baresip/wiki/Guidelines-for-reporting-Issues !