No hookup(answer) event from VoIP module when the peer receives/answers a phone call. (AUD-5207)

[EdwardRyu]

SW/HW versions Audio development kit: ESP32-LyraT Audio kit version : v4.3 Module or chip used: ESP32-WROVER-E IDF version: v4.4.4 ADF version: v2.5

Issue No hookup event from VoIP module when the peer receives/answers a phone call.

Detail I don't see any hook up event defined and supported by VoIP module. Could you help us get the event when the called party answers the call?

Thanks, Edward

[ahhfzhang]

Hi @EdwardRyu ,

The event should be ESP_RTC_EVENT_AUDIO_SESSION_BEGIN, but when the server connects to play the beep or music, this event will not send again.

Thanks.

[EdwardRyu]

Hi Michael,

I have changed the configuration from server side(Twilio) and but media library still sends an event at a wrong time. Here is a detail. If you look at the snippet of the log, you can find that ESP_RTC_EVENT_AUDIO_SESSION_BEGIN event just comes up when (SIP/2.0 183 Session progress) is received.

20 seconds later, the peer users answered the phone and modem received SIP/2.0 200 OK, but no event occurred.

I think ESP_RTC_EVENT_AUDIO_SESSION_BEGIN event should come up after receiving SIP/2.0 200 OK.

Attached log, please have a look into it and then get back to me. newVoIPafterChangingRedirectConfiguration.txt

Thanks, Edward

[2024-02-22 09:30:03.277] D (59721) SIP_SERVICE: _esp_sip_event_handler event 1

[2024-02-22 09:30:15.965] Call-ID: 6F49555B95525C5A2EE1B3D181B8D3B452D899350AD0 [2024-02-22 09:30:15.965] CSeq: 3 INVITE

[2024-02-22 09:30:16.103] Call-ID: 6F49555B95525C5A2EE1B3D181B8D3B452D899350AD0 [2024-02-22 09:30:16.117] CSeq: 3 INVITE

[2024-02-22 09:30:16.272] SIP/2.0 180 Ringing [2024-02-22 09:30:16.272] CSeq: 3 INVITE

[2024-02-22 09:30:16.272] D (72687) SIP_SERVICE: _esp_sip_event_handler event 3 [2024-02-22 09:30:16.272] I (72688) SIP_SERVICE: ESP_RTC_EVENT_CALLING Remote Ring...

[2024-02-22 09:30:18.665] SIP/2.0 183 Session progress [2024-02-22 09:30:18.665] CSeq: 3 INVITE

[2024-02-22 09:30:18.679] D (75144) SIP_SERVICE: _esp_sip_event_handler event 7 [2024-02-22 09:30:18.679] I (75144) SIP_SERVICE: ESP_RTC_EVENT_AUDIO_SESSION_BEGIN

[2024-02-22 09:30:39.216] SIP/2.0 200 OK [2024-02-22 09:30:39.216] CSeq: 3 INVITE

[2024-02-22 09:30:39.229] Call-ID: 6F49555B95525C5A2EE1B3D181B8D3B452D899350AD0 [2024-02-22 09:30:39.229] CSeq: 3 ACK

[ahhfzhang]

Hi Edward,

Got it, I will build a library for you and add callback this 200 event to user.

Thanks.

[ahhfzhang]

Hi Edward,

This esp_media_protocols.zip library you can replace into adf-libs, and you can receive the ESP_RTC_EVENT_ON_CALL event in the user callback.

Thanks.

[EdwardRyu]

Hi Michael,

Now we can receive pickup event at the right time but we have a huge drawback. I have observed two critical problems as follows.

1) Not able to end the call.

• Make an outgoing call from ESP32 to other smartphone.
• Call is connected.
• ESP32 ends the call.
• Call doesn't get terminated.

2) Crash when ending the call in a below procedure.

• Receive and answer the incoming call.
• Call is connected.
• ESP32 ends the call.
• ESP32 gets crashed.

When I look at the size of library, I found that new one is much smaller than I am using as attached screenshot. I wonder if you build the library on top of base line (ESP32, IDFv4.4.4, ADFv2.5) as mentioned at the beginning. We also applied some patches for the issues (issues#991 & issues#1011), but I wonder if they are also applied in the new library.

I will provide the logs later tomorrow, but in the meantime please have a look into this build related issues.

Thanks, Edward

[ahhfzhang]

Hi Edward,

I built a new libesp_media_protocols.zip for (ESP32, IDFv4.4.4, ADFv2.5), but it is still this size. I have tested the two cases you mentioned and it is still ok.

Thanks.

[EdwardRyu]

Hi Michael,

Now it works fine.

Thanks for your support.