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sonuarya / csipsimple

Automatically exported from code.google.com/p/csipsimple
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can't hangup #1900

Closed GoogleCodeExporter closed 9 years ago

GoogleCodeExporter commented 9 years ago 
On outgoing call c sip simple can't initiate hangup. It does not do anything 
pressing hangup, the client does not send anything, examined by tcpdump on 
server side.

phone: LT22i 4.0.4 ics, csipsimple: r1788
server side: asterisk 1.6.2.20

the trace: 

INVITE sip:1231@172.21.57.15 SIP/2.0
Via: SIP/2.0/UDP 
172.21.56.101:43258;rport;branch=z9hG4bKPjfpmYHsiUz81ykYKn2QDA1YbN6QgezTUx
Max-Forwards: 70
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>
Contact: <sip:19@172.21.56.101:43258>
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3241 INVITE
Route: <sip:172.21.57.15;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, 
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_LT22i-15/r1788
Content-Type: application/sdp
Content-Length:   268

v=0
o=- 3554702838 3554702838 IN IP4 172.21.56.101
s=pjmedia
b=AS:84
t=0 0
m=audio 40022 RTP/AVP 8 101
c=IN IP4 172.21.56.101
b=TIAS:64000
a=rtcp:40023 IN IP4 172.21.56.101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 
172.21.56.101:43258;branch=z9hG4bKPjfpmYHsiUz81ykYKn2QDA1YbN6QgezTUx;received=17
2.21.56.101;rport=43258
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>;tag=as5c439d4a
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3241 INVITE
Server: FPBX-2.8.1(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1bbe5660"
Content-Length: 0

ACK sip:1231@172.21.57.15 SIP/2.0
Via: SIP/2.0/UDP 
172.21.56.101:43258;rport;branch=z9hG4bKPjfpmYHsiUz81ykYKn2QDA1YbN6QgezTUx
Max-Forwards: 70
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>;tag=as5c439d4a
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3241 ACK
Route: <sip:172.21.57.15;lr>
Content-Length:  0

INVITE sip:1231@172.21.57.15 SIP/2.0
Via: SIP/2.0/UDP 
172.21.56.101:43258;rport;branch=z9hG4bKPjbQWtvUAyeNOk-SQ.RmSxoShbeB6Cdv0t
Max-Forwards: 70
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>
Contact: <sip:19@172.21.56.101:43258>
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3242 INVITE
Route: <sip:172.21.57.15;lr>
Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, SUBSCRIBE, NOTIFY, REFER, 
MESSAGE, OPTIONS
Supported: replaces, 100rel, timer, norefersub
Session-Expires: 1800
Min-SE: 90
User-Agent: CSipSimple_LT22i-15/r1788
Authorization: Digest username="19", realm="asterisk", nonce="1bbe5660", 
uri="sip:1231@172.21.57.15", response="d7cfbe84578058673fa8123969f89a9c", 
algorithm=MD5
Content-Type: application/sdp
Content-Length:   268

v=0
o=- 3554702838 3554702838 IN IP4 172.21.56.101
s=pjmedia
b=AS:84
t=0 0
m=audio 40022 RTP/AVP 8 101
c=IN IP4 172.21.56.101
b=TIAS:64000
a=rtcp:40023 IN IP4 172.21.56.101
a=sendrecv
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15

SIP/2.0 100 Trying
Via: SIP/2.0/UDP 
172.21.56.101:43258;branch=z9hG4bKPjbQWtvUAyeNOk-SQ.RmSxoShbeB6Cdv0t;received=17
2.21.56.101;rport=43258
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3242 INVITE
Server: FPBX-2.8.1(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:1231@172.21.57.15>
Content-Length: 0

SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 
172.21.56.101:43258;branch=z9hG4bKPjbQWtvUAyeNOk-SQ.RmSxoShbeB6Cdv0t;received=17
2.21.56.101;rport=43258
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>;tag=as000967f1
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3242 INVITE
Server: FPBX-2.8.1(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:1231@172.21.57.15>
Content-Length: 0

SIP/2.0 200 OK
Via: SIP/2.0/UDP 
172.21.56.101:43258;branch=z9hG4bKPjbQWtvUAyeNOk-SQ.RmSxoShbeB6Cdv0t;received=17
2.21.56.101;rport=43258
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>;tag=as000967f1
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3242 INVITE
Server: FPBX-2.8.1(1.6.2.20)
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Contact: <sip:1231@172.21.57.15>
Content-Type: application/sdp
Content-Length: 235

v=0
o=root 224732356 224732356 IN IP4 172.21.57.15
s=Asterisk PBX 1.6.2.20
c=IN IP4 172.21.57.15
t=0 0
m=audio 13570 RTP/AVP 8 101
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=sendrecv

ACK sip:1231@172.21.57.15 SIP/2.0
Via: SIP/2.0/UDP 
172.21.56.101:43258;rport;branch=z9hG4bKPj8X9-b14vYClMWTugHm0og.7LzCdAcKcf
Max-Forwards: 70
From: <sip:19@172.21.57.15>;tag=Z0BRXFPjfGUFIdoHh8khsrWm46QAsael
To: <sip:1231@172.21.57.15>;tag=as000967f1
Call-ID: reppVkPzA23OqieJx4hPQOs1P.MIxPk4
CSeq: 3242 ACK
Content-Length:  0

Original issue reported on code.google.com by attila.d...@gmail.com on 23 Aug 2012 at 9:44

GoogleCodeExporter commented 9 years ago 
Is the client in the same 172.21.57.15 network? Or is it in a different network 
and both network are not routable? (for example 172.21.56.xxx)

Can you collect logs from the app. I think it sends the BYE command, but as a 
subsequent SIP command it doesn't necessarily use the same path to reach the 
other side.
See HowToCollectLogs wiki page.

Original comment by r3gis...@gmail.com on 23 Aug 2012 at 9:48

GoogleCodeExporter commented 9 years ago 
172.21.57.0/24 wired 172.21.56.0/24 wifi, the 2 networks are routed in our 
office.

Original comment by attila.d...@gmail.com on 23 Aug 2012 at 9:52

GoogleCodeExporter commented 9 years ago 
Please forgot the ticket sorry... The juniper srx is stupid.... Sorry again, 
works fine everthing.

Original comment by attila.d...@gmail.com on 23 Aug 2012 at 11:50

GoogleCodeExporter commented 9 years ago 
Ok, Thanks for the update.
No problems :). Don't hesitate if you find something. I learn a lot of things 
helping to debug this kind of situations.

As I rely entierly on pjsip for the sip stack and I'm not a sip expert, my 
skills are often improved when I help to debug some sip problem. Very often 
pjsip does right things as it's widely used by other products and intensively 
test. But understanding why and what configuration should apply the case help 
me to improve the final user app.

Original comment by r3gis...@gmail.com on 23 Aug 2012 at 11:58