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VOIP Example no Audio through ESP after call answer

Posted: Thu Feb 27, 2020 6:08 pm
by Captainavi
Hi,
I have an ESP32 set up with MAX9860 audio codec. A Yate server is running on a local IP that I am using to register the ESP to. I have a softphone Android app that I am using to call the ESP. The softphone app is able to call other softphones on the network and play audio through fine. I am running the most recent VOIP example from ESP-ADF with an output sampling rate of 48Khz. The ESP registers on the network properly as far as I can tell. It will answer calls from the network but no audio plays through once it picks up. It starts the audio codec when the call picks up and plays low frequency noise from the speaker. The noise does not change or react at all to audio from the phone.

Log on the ESP side

Code: Select all

W (43812) SIP: [1970-01-01/00:00:21]<<======================
I (43822) SIP_RTP: rtp local addr=10.1.10.10, port=9999
W (43832) SIP: CHANGE STATE FROM 16, TO 32, :func: sip_uas_process_req:774
E (43832) SIP_RTP: receive task, server addr=10.0.0.50:19014
I (43842) VOIP_EXAMPLE: SIP_EVENT_AUDIO_SESSION_BEGIN
W (106722) SIP: [1970-01-01/00:01:23]<<=====READ 0407 bytes==
I (106722) SIP: 
Log on the Yate server side

Code: Select all

2020-02-27_09:56:53.284099 <sip:NOTE> Registered user '200' expires in 3600 s
2020-02-27_09:57:24.174492 <sip:NOTE> Registered user '100' expires in 180 s
2020-02-27_09:57:25.701446 <sipengine:MILD> Received To tag in 100 answer! (sender bug)
2020-02-27_09:57:26.918878 <sipengine:NOTE> Changing early dialog tag because of forked 200
2020-02-27_09:57:29.920899 <yrtp:WARN> Initial timeout in channel sip/2 wrapper [0x7f9b280152f0]
Does anyone know what these errors mean or what I am doing wrong? The codec is working fine with the other demos so I do not think that it is the problem.

Any thoughts are appreciated.

-Avi