VOIP Example calls don't hang-up with Asteriks (AUD-5592)

[SinnoTogo]

Hello:

Here my setup: esp-idf: V5.1.1 esp-adf: V2.6 Hardware: ESP32-S3-KORVO-2 3.1 PBX: Asterisks running on a rapsberry pi

In general the system works, I can make and receive calls with audio.

But I have problems to hang up calls.

If the call is hang-up by the ESP32 then everything is OK If the call is hang-up by the other end (ie. Microsip or similar) then the ESP32 doesn't hang-up until I explicitly send the esp_rtc_bye command (ie. by ussing a button press on the board).

Here the SIP flow diagram for a case where the end device hangs-up: As you can see the BYE reques is sent by Asteriks but the ESP32 never procesees it.

Next a SIP flow diagra for a case where the ESP32 hangs-up:

In this case the correct sequence is followed and then the call ends properly.

Any idea or hint how to solve the issue?

Thanks!

[TempoTian]

HI I have tested with FreeSwitch when peer send bye, ESP32 can received correctly.
Could you please check on the latest version, just update esp-adf-libs/esp_media_protocols to latest version and try whether it fixed or not? We just commit a solution to fix when resend invite ESP32 response not with SDP may cause server send wrong response. Pay attention latest version rename esp_rtc_init/deinit to esp_rtc_service_init/deinit to avoid conflict with IDF rtc clock API.

[TempoTian]

Here is ESP32 received BYE and response log. You can compare it with your serial log. If possible send me both serial log and related fetched wireshark network packet, I will help to to analysis the possible reasons.

 (53903) SIP: 

BYE sip:1505@192.168.137.27:64339;transport=TCP SIP/2.0
Via: SIP/2.0/TCP 52.81.228.8:9670;rport;branch=z9hG4bKmX9Sv2176Qtgr
Max-Forwards: 70
From: <sip:1509@52.81.228.8:9670>;tag=80099F0aBrtcK
To: <sip:1505@52.81.228.8:9670>;tag=-2029613454
Call-ID: 2A20026AF866CBA21DA7A0E86AFD84A97003AEF7F810
CSeq: 86988519 BYE
User-Agent: FreeSWITCH-mod_sofia/1.10.6-release~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0

I (53961) SIP: [1970-01-01/00:00:26]<<======================
I (53963) SIP_SERVICE: ESP_RTC_EVENT_AUDIO_SESSION_END
W (53974) AUDIO_PIPELINE: There are no listener registered
I (53975) AUDIO_PIPELINE: audio_pipeline_unlinked
I (53977) AV_STREAM: _audio_dec task stoped
I (53988) SIP_SERVICE: ESP_RTC_EVENT_HANGUP
W (53989) SIP: CHANGE STATE FROM 96, TO 2, :func: sip_uas_process_req:1043
I (54000) SIP: [1970-01-01/00:00:26]=======WRITE 0418 bytes>>
I (54001) SIP: 

SIP/2.0 200 OK
Via: SIP/2.0/TCP 52.81.228.8:9670;rport;branch=z9hG4bKmX9Sv2176Qtgr
Contact: <sip:1505@192.168.137.27:64339;transport=TCP>
From: <sip:1509@52.81.228.8:9670>;tag=80099F0aBrtcK
To: <sip:1505@52.81.228.8:9670>;tag=-2029613454
Call-ID: 2A20026AF866CBA21DA7A0E86AFD84A97003AEF7F810
CSeq: 86988519 BYE
Server: ESP32 SIP/2.0
Allow: ACK, INVITE, BYE, UPDATE, CANCEL, OPTIONS, INFO
Content-Length: 0