I found the bug in my code and now video works! But only in one direction.
If I call from Android to Inosmart then video works. If Inosmart call Android I get only audio.
What should be the problem? I also test both my clients without Asterisk, using sip.linphone.org and video works as espected…
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Executing [android@phones:1] NoOp(“SIP/inosmart-00000017”, “Calling android”) in new stack
– Executing [android@phones:2] Dial(“SIP/inosmart-00000017”, “SIP/android”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
– Called SIP/android
– Registered SIP ‘inosmart’ at 192.168.2.111:5071
– SIP/android-00000018 is ringing
raspberrypi*CLI> sip set debug on
SIP Debugging enabled
<— SIP read from UDP:192.168.2.143:5063 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.94:5060;branch=z9hG4bK65340812
Contact: sip:android@192.168.2.143:5063;+sip.instance="urn:uuid:3F33D001-F76E-CB0D-2029-CB403369F158"
To: sip:android@192.168.2.143:5063;tag=db7b8877
From: sip:inosmart@192.168.2.94;tag=as35489c8d
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 102 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, outbound, path
User-Agent: PortSIP SDK for Android
Allow-Events: hold, talk, conference
Content-Length: 290
v=0
o=- 1512930115 1 IN IP4 192.168.2.143
s=portsip.com
c=IN IP4 192.168.2.143
t=0 0
m=audio 20022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 43022 RTP/AVP 96
a=rtpmap:96 VP8/90000
a=sendrecv
<------------->
> 0x2b15498 – Probation passed - setting RTP source address to 192.168.2.143:43022
— (13 headers 14 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found video description format VP8 for ID 96
Capabilities: us - (ulaw|alaw|vp8), peer - audio=(ulaw|alaw)/video=(vp8)/text=(nothing), combined - (ulaw|alaw|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.143:20022
Peer video RTP is at port 192.168.2.143:43022
sip_route_dump: route/path hop: sip:android@192.168.2.143:5063
set_destination: Parsing sip:android@192.168.2.143:5063 for address/port to send to
set_destination: set destination to 192.168.2.143:5063
Transmitting (no NAT) to 192.168.2.143:5063:
ACK sip:android@192.168.2.143:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.94:5060;branch=z9hG4bK19728ddc
Max-Forwards: 70
From: sip:inosmart@192.168.2.94;tag=as35489c8d
To: sip:android@192.168.2.143:5063;tag=db7b8877
Contact: sip:inosmart@192.168.2.94:5060
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.15.0
Content-Length: 0
-- SIP/android-00000018 answered SIP/inosmart-00000017
Audio is at 12628
Video is at 192.168.2.111:18624
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec vp8 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<— Reliably Transmitting (no NAT) to 192.168.2.111:5071 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.111:5071;branch=z9hG4bK.92fKyEjud;received=192.168.2.111;rport=5071
From: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
To: sip:android@192.168.2.111;tag=as0873c44e
Call-ID: ZBJP-YhNwi
CSeq: 21 INVITE
Server: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:android@192.168.2.111:5060
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 806815063 806815063 IN IP4 192.168.2.111
s=Asterisk PBX 13.15.0
c=IN IP4 192.168.2.111
b=CT:384
t=0 0
m=audio 12628 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 18624 RTP/AVP 96
a=rtpmap:96 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv
<------------>
– Channel SIP/android-00000018 joined ‘simple_bridge’ basic-bridge
– Channel SIP/inosmart-00000017 joined ‘simple_bridge’ basic-bridge
> Bridge fa640dbc-0b75-4281-aa04-58bc742dcc8c: switching from simple_bridge technology to native_rtp
set_destination: Parsing sip:android@192.168.2.143:5063 for address/port to send to
set_destination: set destination to 192.168.2.143:5063
Audio is at 18488
Video is at 192.168.2.111:9078
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec vp8 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.143:5063:
INVITE sip:android@192.168.2.143:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.94:5060;branch=z9hG4bK0a9ff206
Max-Forwards: 70
From: sip:inosmart@192.168.2.94;tag=as35489c8d
To: sip:android@192.168.2.143:5063;tag=db7b8877
Contact: sip:inosmart@192.168.2.94:5060
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 354
v=0
o=root 334558956 334558957 IN IP4 192.168.2.111
s=Asterisk PBX 13.15.0
c=IN IP4 192.168.2.111
b=CT:384
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 9078 RTP/AVP 96
a=rtpmap:96 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv
> Remotely bridged 'SIP/inosmart-00000017' and 'SIP/android-00000018' - media will flow directly between them
> Remotely bridged 'SIP/inosmart-00000017' and 'SIP/android-00000018' - media will flow directly between them
<— SIP read from UDP:192.168.2.111:5071 —>
ACK sip:android@192.168.2.111:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.111:5071;rport;branch=z9hG4bK.W–t7AtLs
From: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
To: sip:android@192.168.2.111;tag=as0873c44e
CSeq: 21 ACK
Call-ID: ZBJP-YhNwi
Max-Forwards: 70
Authorization: Digest realm=“asterisk”, nonce=“54a67460”, algorithm=MD5, username=“inosmart”, uri="sip:android@192.168.2.111", response=“8ab33b02532909b69fc97e4b42a62c65”
<------------->
— (8 headers 0 lines) —
set_destination: Parsing sip:inosmart@192.168.2.111:5071 for address/port to send to
set_destination: set destination to 192.168.2.111:5071
Audio is at 12628
Video is at 192.168.2.143:43022
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec vp8 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.111:5071:
INVITE sip:inosmart@192.168.2.111:5071 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK68b9c75d;rport
Max-Forwards: 70
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Contact: sip:android@192.168.2.111:5060
Call-ID: ZBJP-YhNwi
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 806815063 806815064 IN IP4 192.168.2.143
s=Asterisk PBX 13.15.0
c=IN IP4 192.168.2.143
b=CT:384
t=0 0
m=audio 20022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 43022 RTP/AVP 96
a=rtpmap:96 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv
<— SIP read from UDP:192.168.2.143:5063 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.94:5060;branch=z9hG4bK0a9ff206
To: sip:android@192.168.2.143:5063;tag=db7b8877
From: sip:inosmart@192.168.2.94;tag=as35489c8d
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 103 INVITE
Content-Length: 0
<------------->
— (7 headers 0 lines) —
> 0x75f195e0 – Probation passed - setting RTP source address to 192.168.2.111:7078
<— SIP read from UDP:192.168.2.111:5071 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK68b9c75d;rport
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Call-ID: ZBJP-YhNwi
CSeq: 102 INVITE
<------------->
— (6 headers 0 lines) —
<— SIP read from UDP:192.168.2.111:5071 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK68b9c75d;rport
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Call-ID: ZBJP-YhNwi
CSeq: 102 INVITE
User-Agent: (belle-sip/1.4.2)
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: “ino” sip:inosmart@192.168.2.111:5071;+sip.instance="urn:uuid:ba3a9e4c-ff1b-4a19-8d29-bf7df6d51407"
Content-Type: application/sdp
Content-Length: 200
v=0
o=inosmart 3119 3879 IN IP4 192.168.2.111
s=Talk
c=IN IP4 192.168.2.111
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 96
a=rtpmap:96 VP8/90000
<------------->
— (12 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found video description format VP8 for ID 96
Capabilities: us - (ulaw|alaw|vp8), peer - audio=(ulaw|alaw)/video=(vp8)/text=(nothing), combined - (ulaw|alaw|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.111:7078
Peer video RTP is at port 192.168.2.111:9078
set_destination: Parsing sip:inosmart@192.168.2.111:5071 for address/port to send to
set_destination: set destination to 192.168.2.111:5071
Transmitting (no NAT) to 192.168.2.111:5071:
ACK sip:inosmart@192.168.2.111:5071 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK0415665a;rport
Max-Forwards: 70
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Contact: sip:android@192.168.2.111:5060
Call-ID: ZBJP-YhNwi
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.15.0
Content-Length: 0
> 0x75f195e0 -- Probation passed - setting RTP source address to 192.168.2.111:7078
<— SIP read from UDP:192.168.2.143:5063 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.94:5060;branch=z9hG4bK0a9ff206
Contact: sip:android@192.168.2.143:5063;+sip.instance="urn:uuid:3F33D001-F76E-CB0D-2029-CB403369F158"
To: sip:android@192.168.2.143:5063;tag=db7b8877
From: sip:inosmart@192.168.2.94;tag=as35489c8d
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 103 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, outbound, path
User-Agent: PortSIP SDK for Android
Allow-Events: hold, talk, conference
Content-Length: 290
v=0
o=- 1512930115 2 IN IP4 192.168.2.143
s=portsip.com
c=IN IP4 192.168.2.143
t=0 0
m=audio 20022 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
m=video 43022 RTP/AVP 96
a=rtpmap:96 VP8/90000
a=sendrecv
<------------->
— (13 headers 14 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found video description format VP8 for ID 96
Capabilities: us - (ulaw|alaw|vp8), peer - audio=(ulaw|alaw)/video=(vp8)/text=(nothing), combined - (ulaw|alaw|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.143:20022
Peer video RTP is at port 192.168.2.143:43022
set_destination: Parsing sip:android@192.168.2.143:5063 for address/port to send to
set_destination: set destination to 192.168.2.143:5063
Transmitting (no NAT) to 192.168.2.143:5063:
ACK sip:android@192.168.2.143:5063 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.94:5060;branch=z9hG4bK03b26dc3
Max-Forwards: 70
From: sip:inosmart@192.168.2.94;tag=as35489c8d
To: sip:android@192.168.2.143:5063;tag=db7b8877
Contact: sip:inosmart@192.168.2.94:5060
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.15.0
Content-Length: 0
<— SIP read from UDP:192.168.2.111:5071 —>
<------------->
<— SIP read from UDP:192.168.2.143:5063 —>
BYE sip:inosmart@192.168.2.94:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.143:5063;branch=z9hG4bK-524287-1—8905af18eab2fb16;rport
Max-Forwards: 70
Contact: sip:android@192.168.2.143:5063;+sip.instance="urn:uuid:3F33D001-F76E-CB0D-2029-CB403369F158"
To: sip:inosmart@192.168.2.94;tag=as35489c8d
From: sip:android@192.168.2.143:5063;tag=db7b8877
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 2 BYE
User-Agent: PortSIP SDK for Android
Content-Length: 0
<------------->
— (10 headers 0 lines) —
Sending to 192.168.2.143:5063 (no NAT)
Scheduling destruction of SIP dialog ‘43af25057587066b2437f02f3499ac25@192.168.2.94:5060’ in 6400 ms (Method: BYE)
<— Transmitting (no NAT) to 192.168.2.143:5063 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.143:5063;branch=z9hG4bK-524287-1—8905af18eab2fb16;received=192.168.2.143;rport=5063
From: sip:android@192.168.2.143:5063;tag=db7b8877
To: sip:inosmart@192.168.2.94;tag=as35489c8d
Call-ID: 43af25057587066b2437f02f3499ac25@192.168.2.94:5060
CSeq: 2 BYE
Server: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0
<------------>
– Channel SIP/android-00000018 left ‘native_rtp’ basic-bridge
set_destination: Parsing sip:inosmart@192.168.2.111:5071 for address/port to send to
set_destination: set destination to 192.168.2.111:5071
Audio is at 12628
Video is at 192.168.2.111:18624
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding video codec vp8 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.2.111:5071:
INVITE sip:inosmart@192.168.2.111:5071 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK4fcf8594;rport
Max-Forwards: 70
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Contact: sip:android@192.168.2.111:5060
Call-ID: ZBJP-YhNwi
CSeq: 103 INVITE
User-Agent: Asterisk PBX 13.15.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
X-asterisk-Info: SIP re-invite (External RTP bridge)
Content-Type: application/sdp
Content-Length: 356
v=0
o=root 806815063 806815065 IN IP4 192.168.2.111
s=Asterisk PBX 13.15.0
c=IN IP4 192.168.2.111
b=CT:384
t=0 0
m=audio 12628 RTP/AVP 0 8 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 18624 RTP/AVP 96
a=rtpmap:96 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv
-- Channel SIP/inosmart-00000017 left 'native_rtp' basic-bridge <fa640dbc-0b75-4281-aa04-58bc742dcc8c>
== Spawn extension (phones, android, 2) exited non-zero on 'SIP/inosmart-00000017’
Scheduling destruction of SIP dialog ‘ZBJP-YhNwi’ in 32000 ms (Method: ACK)
<— SIP read from UDP:192.168.2.111:5071 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK4fcf8594;rport
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Call-ID: ZBJP-YhNwi
CSeq: 103 INVITE
<------------->
— (6 headers 0 lines) —
<— SIP read from UDP:192.168.2.111:5071 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK4fcf8594;rport
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Call-ID: ZBJP-YhNwi
CSeq: 103 INVITE
User-Agent: (belle-sip/1.4.2)
Supported: outbound
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE
Contact: “ino” sip:inosmart@192.168.2.111:5071;+sip.instance="urn:uuid:ba3a9e4c-ff1b-4a19-8d29-bf7df6d51407"
Content-Type: application/sdp
Content-Length: 200
v=0
o=inosmart 3119 3881 IN IP4 192.168.2.111
s=Talk
c=IN IP4 192.168.2.111
t=0 0
m=audio 7078 RTP/AVP 0 8 101
a=rtpmap:101 telephone-event/8000
m=video 9078 RTP/AVP 96
a=rtpmap:96 VP8/90000
<------------->
— (12 headers 9 lines) —
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format telephone-event for ID 101
Found RTP video format 96
Found video description format VP8 for ID 96
Capabilities: us - (ulaw|alaw|vp8), peer - audio=(ulaw|alaw)/video=(vp8)/text=(nothing), combined - (ulaw|alaw|vp8)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.2.111:7078
Peer video RTP is at port 192.168.2.111:9078
set_destination: Parsing sip:inosmart@192.168.2.111:5071 for address/port to send to
set_destination: set destination to 192.168.2.111:5071
Transmitting (no NAT) to 192.168.2.111:5071:
ACK sip:inosmart@192.168.2.111:5071 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK34bc5385;rport
Max-Forwards: 70
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Contact: sip:android@192.168.2.111:5060
Call-ID: ZBJP-YhNwi
CSeq: 103 ACK
User-Agent: Asterisk PBX 13.15.0
Content-Length: 0
set_destination: Parsing sip:inosmart@192.168.2.111:5071 for address/port to send to
set_destination: set destination to 192.168.2.111:5071
Reliably Transmitting (no NAT) to 192.168.2.111:5071:
BYE sip:inosmart@192.168.2.111:5071 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK2e1ca7dc;rport
Max-Forwards: 70
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Call-ID: ZBJP-YhNwi
CSeq: 104 BYE
User-Agent: Asterisk PBX 13.15.0
Proxy-Authorization: Digest username=“inosmart”, realm=“asterisk”, algorithm=MD5, uri=“sip:192.168.2.111”, nonce=“54a67460”, response="733822716dee23e8f95769ccd434627c"
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0
Scheduling destruction of SIP dialog ‘ZBJP-YhNwi’ in 32000 ms (Method: ACK)
<— SIP read from UDP:192.168.2.111:5071 —>
SIP/2.0 200 Ok
Via: SIP/2.0/UDP 192.168.2.111:5060;branch=z9hG4bK2e1ca7dc;rport
From: sip:android@192.168.2.111;tag=as0873c44e
To: sip:inosmart@192.168.2.111;tag=aLhMjSY8Y
Call-ID: ZBJP-YhNwi
CSeq: 104 BYE
User-Agent: (belle-sip/1.4.2)
Supported: outbound