Webrtc to SIP fails . Flashphoner cancels the ca

[mbedial]

Hi friends,
although we have our Flashphoner Web Call Server working for a long time, we have configured a new scenario and it fails.
To be exact, we use a webrtc client (android) that calls to the WCS. WCS gateway converts the call to SIP that is sent to asterisk PBX and we have this signalling

WCS ------------ INVITE --------------> Asterisk
WCS <---------100 trying -------------- Asterisk
WCS ------------- CANCEL-------------> Asterisk

The details of that cancel is here

08:47:18,690 INFO WCS4CallbackHandler - pool-22-thread-3 notifyBalance 918883334@hasip.custofake.com
08:47:19,507 INFO WCS4Handler - API-ASYNC-pool-21-thread-8162 hangup callId: dbcbacb3-8ccc-46c4-8325-afdac1cd8fcd client: ClientConfig{rtspMedia=false, wsTunnel=false, login='918883334', authenticationName='918883334', password='4488112', domain='hasip.custofake.com', outboundProxy='10.0.0.4', localPort=40004, port=5080, visibleName='918883334', regRequired=true, applicationName='null', swfUrl='null', useProxy=true, qValue='null', pAssociatedUri='', contactParams='null', authDate=Tue Feb 20 08:47:02 CET 2018, authToken='/42.35.186.108:49097/85.63.27.22:9443', logsFolderName='null', clientVersion='null', clientOSVersion='null', clientBrowserVersion='null', wsTunnelPacketization2='false', custom='{custom={}, origin=null}'}
08:47:19,508 DEBUG sipMessages - API-ASYNC-pool-21-thread-8162

--------------------> CANCEL sip:6988896@hasip.custofake.com SIP/2.0
from: /10.0.0.4:40004
to: /10.0.0.4:5080
time: 1519112839508
timeStamp:
isSender: true
transactionId: z9hg4bked8bf3bec572ff8023e7212855fba0af:cancel
callId: dbcbacb3-8ccc-46c4-8325-afdac1cd8fcd

CANCEL sip:6988896@hasip.custofake.com SIP/2.0
Call-ID: dbcbacb3-8ccc-46c4-8325-afdac1cd8fcd
To: <sip:6988896@hasip.custofake.com>
CSeq: 2 CANCEL
From: "918883334" <sip:918883334@hasip.custofake.com>;tag=d0f8570f
Via: SIP/2.0/UDP 10.0.0.4:40004;branch=z9hG4bKed8bf3bec572ff8023e7212855fba0af
Max-Forwards: 70
Content-Length: 0


08:47:19,508 INFO SipCallProcessor - API-ASYNC-pool-21-thread-8162 terminate: sipCall.id=dbcbacb3-8ccc-46c4-8325-afdac1cd8fcd
08:47:19,508 INFO SipCallProcessor - API-ASYNC-pool-21-thread-8162 Stop rtc media session id RTC-dbcbacb3-8ccc-46c4-8325-afdac1cd8fcd, sipCallId dbcbacb3-8ccc-46c4-8325-afdac1cd8fcd
08:47:19,508 INFO Agent - API-ASYNC-pool-21-thread-8162 ICE state changed from Waiting to Terminated. Local ufrag RTC-dbcbacb3-8ccc-46c4-8325-afdac1cd8fcdckp091c6p2sngl
08:47:19,508 DEBUG sipMessages - UDPMessageChannelThread-40004-62463


See the attached file with the entire log
As I said, we have similar scenarios that connects with asterisk ann they work without problems.

Thanks in advance for your help.

Mario

 

[Max]

Hello
Form the logs we can see that Android App does call.hangup()
Try to establish websocket connection (not ssl)

ws://host:8080

then make a dump on WCS end

tcpdump tcp port 8080 -s 4096 -w log.pcap

Most likely you will see that client-side sends 'hangup' message to WCS server.
Try to localize this hangup invocation in your application code.
Try to compare this behavior with our default application Phone for Android.