Hi,
I had tried jssip and got same result.
Bye signal could be received in audio calls but not in video
calls.
Here is what I found (both sipml5 and jssip.)
When client(U1109903) hangup video call, the BYE signal would
reach to FS but not to the other WebRTC client.
Then the other WebRTC cleint(U1109902) hangup the call, its Chrome
log shows all BYE signals are received in this time.
========================
BYE
sip:U1109903@10.0.199.33:5060;transport=udp SIP/2.0
Via: SIP/2.0/WS l1ulafktoii1.invalid;branch=z9hG4bK9006154
Max-Forwards: 69
To:
<sip:U1109903@10.0.199.33>;tag=jmB2eBSH0Bggm
From: "U1109902"
<sip:U1109902@10.0.199.33>;tag=ejrm0q06qe
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 2213 BYE
Supported: path, outbound, gruu
User-Agent: JsSIP 0.3.7
Content-Length: 0
...
BYE
sip:7pmon84c@l1ulafktoii1.invalid;transport=ws;ob SIP/2.0
Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKtBSFaU4mScSZp
Max-Forwards: 70
From:
<sip:U1109903@10.0.199.33>;tag=jmB2eBSH0Bggm
To: "U1109902"
<sip:U1109902@10.0.199.33>;tag=ejrm0q06qe
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 53082708 BYE
Contact:
<sip:U1109903@10.0.199.33:5060;transport=udp>
User-Agent:
FreeSWITCH-mod_sofia/1.5.8b+git~20131205T011809Z~4a2054a951~64bit
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE,
REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE
Supported: timer, precondition, path, replaces
Reason: Q.850;cause=16;text="NORMAL_CLEARING"
Content-Length: 0
jssip-devel.js:671
JsSIP | TRANSPORT | sending WebSocket message:
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/WS 10.0.199.33:5066;branch=z9hG4bKtBSFaU4mScSZp
To: "U1109902"
<sip:U1109902@10.0.199.33>;tag=ejrm0q06qe
From:
<sip:U1109903@10.0.199.33>;tag=jmB2eBSH0Bggm
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 53082708 BYE
Content-Length: 0
jssip-devel.js:520
JsSIP | TRANSACTION | Timer J expired for non-INVITE server
transaction z9hG4bKtBSFaU4mScSZp jssip-devel.js:2073
JsSIP | TRANSPORT | received WebSocket text message:
SIP/2.0 200 OK
Via: SIP/2.0/WS
l1ulafktoii1.invalid;branch=z9hG4bK9006154;received=10.0.1.69;rport=53126
From: "U1109902"
<sip:U1109902@10.0.199.33>;tag=ejrm0q06qe
To:
<sip:U1109903@10.0.199.33>;tag=jmB2eBSH0Bggm
Call-ID: 3gc45bilg4rbt647ue0u
CSeq: 2213 BYE
Content-Length: 0
========================
於 2013/12/12 下午 02:47, James Mortensen 提到: