Incoming call accepted, Status = call!

[indotronic]

if incoming call accepted, cannot get cancel if caller hang up the call.

[rousir]

If the call is answered, send call hang up command to hang up the phone

void SipMachine::bye()
{
    sipBye();
    status = idle;
}

If the caller cancels the call, esp32 will reset the state after receiving the bye command

ArduinoVoip/src/SipHeader.cpp

Line 40 in d09927c

if (in.indexOf("BYE sip:") > -1)

ArduinoVoip/src/SipMachine.cpp

Line 283 in d09927c

case 1:

[indotronic]

If the caller cancels the call :

ArduinoVoip/src/SipMachine.cpp

Line 122 in d09927c

if ((status == call) & (timeStOut % 20000 == 0))

if caller calling and answered, speak, and caller hang up. Esp no get respondcode, still lopoping in here!

[indotronic]

and if time experied the sip cannot re-registering!
in debug show
07:27:39.221 -> sip server disconnected, reconnect...
07:27:39.256 -> Connect SIP Client on IP 192.168.43.202 Port 5060 ret= true
07:27:39.256 -> ***** Register *****
07:27:39.256 -> REGISTER sip:192.168.43.202 SIP/2.0
07:27:39.256 -> Via: SIP/2.0/TCP 192.168.43.54:5060;branch=z9hG4bK-0Oi2Qc1Kf6Od3Ws5Sq4Pf8Tw6We5Kx
07:27:39.256 -> Max-Forwards: 70
07:27:39.290 -> From: sip:[email protected]:5060;tag=
07:27:39.290 -> To: sip:[email protected]:5060;tag=as7c2038dd
07:27:39.290 -> Call-ID: [email protected]:5060
07:27:39.290 -> CSeq: 1 REGISTER
07:27:39.290 -> User-Agent: esp32 sip
07:27:39.290 -> Contact: sip:[email protected];transport=TCP
07:27:39.290 -> Allow: INVITE,ACK,OPTIONS,BYE,CANCEL,SUBSCRIBE,NOTIFY,REFER,MESSAGE,INFO,PING
07:27:39.290 -> Content-Length: 0

[rousir]

Okay, I didn’t encounter this problem during the test. I have been busy recently. I will check it when I have time.
The sip protocol is simple, you can use the debug information to check that the step was unsuccessful

[indotronic]

Ok,
fix now for re registering issues!

pls see the code

ArduinoVoip/src/SipMachine.cpp

Line 190 in d09927c

if (sipHeader.contentLength < 0)

its on if condition
if (sipHeader.contentLength > 0)
{

}

its work?

[rousir]

if (sipHeader.contentLength > 0)
{
    sipHeader.contentLength -= (str.length() + 1);
    debugL1_println(str.c_str());
    parserSdp(str);

    if (sipHeader.contentLength < 0)
    {
        exec();

        if (status == messageIn) {
            debugL1_println(str.c_str());
            messageData = str;
        }
    }
}

It works and is used to detect the content of the sip body

[indotronic]

If incoming call show log it

19:08:37.381 -> 192.168.43.54
19:08:37.381 -> Init i2s amp...
19:08:37.719 -> Respons Code = 401
19:08:38.026 -> Respons Code = 401
19:08:38.225 -> Respons Code = 200
19:08:38.225 -> REGISTER OK
19:08:52.141 -> Respons Code = 200
19:08:52.141 -> REGISTER OK
19:09:07.408 -> Respons Code = 2
19:09:07.408 -> Ringing Call Nr. 2013
19:09:12.389 -> Accept incoming Call 2013
19:09:12.389 -> Accept call
19:09:12.422 -> Accept call
19:09:12.725 -> Accept call
19:09:12.725 -> Accept call
19:09:12.725 -> Accept call
19:09:12.725 -> Respons Code = 0
19:09:13.361 -> Respons Code = 2
19:09:13.361 -> Ringing Call Nr. 2013
19:09:18.379 -> Accept incoming Call 2013
19:09:18.379 -> Accept call
19:09:18.379 -> Accept call
19:09:18.379 -> Accept call
19:09:18.379 -> Accept call
19:09:18.379 -> Accept call
19:09:18.379 -> Accept call
19:09:18.447 -> Accept call
19:09:18.447 -> Accept call
19:09:18.447 -> Respons Code = 0
19:09:18.547 -> Respons Code = 1
19:09:18.547 -> 2013 Bye Call

[rousir]

After receiving the call, esp needs to send the answer command many times to answer it normally?
收到呼叫后，esp需要发很多次接听命令才能正常接听？

[indotronic]

i think esp no send command many time..
and auto hangup after 10sec accept call

[indotronic]

the call problem ended outomatic after 10 second solved.. but if caller hang up, caller not send BYE..

[indotronic]

if make call not work, just show

***** Invite *****
15:39:32.035 -> INVITE sip:[email protected] SIP/2.0
15:39:32.035 -> Via: SIP/2.0/TCP 192.168.43.202;branch=z9hG4bK-4Hs5Jq4Vp4Vk2Xk0Sm1Bv1Mc2Bc8Ri;rport;alias
15:39:32.035 -> To: sip:[email protected]
15:39:32.035 -> From: sip:[email protected];tag=
15:39:32.035 -> Call-ID: [email protected]
15:39:32.035 -> CSeq: 1 INVITE
15:39:32.035 -> Contact: sip:[email protected];transport=TCP
15:39:32.035 -> User-Agent: esp32 sip
15:39:32.035 -> Supported: replaces
15:39:32.035 -> Allow: ACK,PRACK,BYE,CANCEL,INVITE,UPDATE,MESSAGE,INFO,OPTIONS,SUBSCRIBE,NOTIFY,REFER
15:39:32.070 -> Allow-Events: presence,refer,telephone-event,keep-alive,dialog
15:39:32.070 -> Accept: application/sdp,application/dtmf-relay,text/plain
15:39:32.070 -> Content-Type: application/sdp
15:39:32.070 -> Content-Length: 182
15:39:32.070 ->
15:39:32.070 -> v=0
15:39:32.070 -> o=2012 65485742 50506087 IN IP4 192.168.43.54
15:39:32.070 -> s=esp32 sip
15:39:32.070 -> c=IN IP4 192.168.43.54
15:39:32.070 -> t=0 0
15:39:32.070 -> m=audio 64389 RTP/AVP 8 127
15:39:32.070 -> a=rtpmap:8 PCMA/8000
15:39:32.070 -> a=rtpmap:127 telephone-event/8000
15:39:32.104 ->