I guess that you test by calling another user extension under the same account, right? If that is the case, can you test be making an outbound call to your mobile phone number (hope that you are in the U.S so you can make such local call.)
Yes, I was calling by extension under the same company account.
I tested the same code with a mobile phone number it works in the same way - the answered event comes immediately after the call was created.
It is a known issue if the callee is a RingCentral number, it will be answered immediately. It is by design. I don't like the behavior but I cannot change it. It's not due to the softphone SDK, it is a RingCentral server side behavior.
If you make a call to your own number (not RingCentral number). Make sure the callee is not a RingCentral number, and in theory, this issue will not be reproduced. I just tried my own mobile phone and it works as expected.
You can see that "SIP/2.0 183 Session Progress" at 10:46:47 and "SIP/2.0 200 OK" at 10:47:02. The 5 seconds difference is how long it took me to answer the phone call.
Receiving...(Wed Apr 17 2024 10:46:47 GMT-0700 (Pacific Daylight Time))
SIP/2.0 100 Trying
Via: SIP/2.0/TCP 00c8257b-c28c-4f31-a30a-07f998453a13.invalid;branch=z9hG4bK-9b592234-18dd-434b-8de0-206fea4788a1;received=45.62.187.76
To: <sip:secret@sip.ringcentral.com>;tag=44b438c2-3b81-40a7-a9bb-ba992227049e
From: <sip:secret@sip.ringcentral.com>;tag=886839d9-a9cd-4aff-bd62-60d4cae3afeb
Call-Id: d370268b-d19d-466a-9bb6-294fc5a22868
CSeq: 2537 INVITE
Content-Length: 0
Receiving...(Wed Apr 17 2024 10:46:47 GMT-0700 (Pacific Daylight Time))
SIP/2.0 183 Session Progress
Via: SIP/2.0/TCP 00c8257b-c28c-4f31-a30a-07f998453a13.invalid;branch=z9hG4bK-9b592234-18dd-434b-8de0-206fea4788a1;received=45.62.187.76
To: <sip:secret@sip.ringcentral.com>;tag=10.13.23.183-5070-79494a5d-9b60-41ad
From: <sip:secret@sip.ringcentral.com>;tag=886839d9-a9cd-4aff-bd62-60d4cae3afeb
Call-Id: d370268b-d19d-466a-9bb6-294fc5a22868
CSeq: 2537 INVITE
Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO
Content-Type: application/sdp
Contact: <sip:16504306662@104.245.57.195:5091;transport=tcp>
p-rc-api-ids: party-id=p-a0d17b7307ffbz18eed2b16fcz47c8ad0000-1;session-id=s-a0d17b7307ffbz18eed2b16fcz47c8ad0000
Content-Length: 225
v=0
o=- 1248600857100441966 8491060242449348806 IN IP4 208.87.41.71
s=SmcSip
c=IN IP4 208.87.41.71
t=0 0
m=audio 27920 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
Receiving...(Wed Apr 17 2024 10:47:02 GMT-0700 (Pacific Daylight Time))
SIP/2.0 200 OK
Via: SIP/2.0/TCP 00c8257b-c28c-4f31-a30a-07f998453a13.invalid;branch=z9hG4bK-9b592234-18dd-434b-8de0-206fea4788a1;received=45.62.187.76
To: <sip:secret@sip.ringcentral.com>;tag=10.13.23.183-5070-79494a5d-9b60-41ad
From: <sip:secret@sip.ringcentral.com>;tag=886839d9-a9cd-4aff-bd62-60d4cae3afeb
Call-Id: d370268b-d19d-466a-9bb6-294fc5a22868
CSeq: 2537 INVITE
Supported: replaces, timer, diversion, histinfo
Allow: SUBSCRIBE, NOTIFY, REFER, INVITE, ACK, BYE, CANCEL, UPDATE, INFO
Content-Type: application/sdp
Contact: <sip:16504306662@104.245.57.195:5091;transport=tcp>
p-rc-api-ids: party-id=p-a0d17b7307ffbz18eed2b16fcz47c8ad0000-1;session-id=s-a0d17b7307ffbz18eed2b16fcz47c8ad0000
Content-Length: 225
v=0
o=- 1248600857100441966 8491060242449348806 IN IP4 208.87.41.71
s=SmcSip
c=IN IP4 208.87.41.71
t=0 0
m=audio 27920 RTP/AVP 0 101
a=sendrecv
a=rtpmap:0 pcmu/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
But we need this to create a call inside the company and stream the audio here.
Is there any other way to receive an event, maybe some custom event, when the call is answered?
Thanks.
