Asterisk 13 not able to make video call

Hi,
after I installed Asterisk 13 I can not make video call. Voice is working on both side, but video does not streaming. I am using PCMU,PCMA and VP8 video codec.
Video should stream in only one way - from Inosmart to Android.

Any suggestions?

Thanks
Bostjan

sip.conf
[general]
context=public
allowoverlap=no
tcpenable=no
tcpbindaddr=0.0.0.0
transport=udp
srvlookup=yes
bindport=5060
videosupport=yes
accept_outofcall_message=yes
outofcall_message_contex=inomsg
auth_message_request=no

[inosmart]
type=friend
context=phones
secret=12345678
qualify=yes
host=dynamic
disallow=all
allow=ulaw,alaw,vp8

[android]
type=friend
context=phones
secret=12345678
qualify=yes
host=dynamic
disallow=all
allow=ulaw,alaw,vp8

extensions.conf
[phones]
exten => android,1,NoOp(Klicem android)
same => n,Dial(SIP/android)
same => n,HangUp

exten => inosmart,1,NoOp(Klicem inosmart))
same => n,Dial(SIP/inosmart)
same => n,HangUp

[inomsg]
exten => _XXX,1,Set(ACTUALTO=${CUT(MESSAGE(to),@,1)})
exten => _XXX,n,MessageSend(${ACTUALTO},${MESSAGE(from)})
exten => _XXX,n,Hangup

Asterisk Debug:
<------------>
– Executing [inosmart@phones:1] NoOp(“SIP/android-00000034”, “Calling inosmart)”) in new stack
– Executing [inosmart@phones:2] Dial(“SIP/android-00000034”, “SIP/inosmart”) in new stack
== Using SIP VIDEO CoS mark 6
== Using SIP RTP CoS mark 5
Audio is at 17294
Video is at 192.168.2.111:10476
Adding codec ulaw to SDP
Adding video codec vp8 to SDP
Adding codec alaw 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=z9hG4bK13abc69f
Max-Forwards: 70
From: “android” sip:android@192.168.2.111;tag=as27d05a17
To: sip:inosmart@192.168.2.111:5071
Contact: sip:android@192.168.2.111:5060
Call-ID: 7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 13.18.3
Date: Sun, 10 Dec 2017 10:50:22 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 358

v=0
o=root 444720569 444720569 IN IP4 192.168.2.111
s=Asterisk PBX 13.18.3
c=IN IP4 192.168.2.111
b=CT:384
t=0 0
m=audio 17294 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 10476 RTP/AVP 120
a=rtpmap:120 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv


-- Called SIP/inosmart

<— 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=z9hG4bK13abc69f
From: “android” sip:android@192.168.2.111;tag=as27d05a17
To: sip:inosmart@192.168.2.111:5071
Call-ID: 7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060
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=z9hG4bK13abc69f
From: “android” sip:android@192.168.2.111;tag=as27d05a17
To: sip:inosmart@192.168.2.111:5071;tag=jz-b0NY
Call-ID: 7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060
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: sip:inosmart@192.168.2.111:5071;+sip.instance="urn:uuid:ba3a9e4c-ff1b-4a19-8d29-bf7df6d51407"
Content-Type: application/sdp
Content-Length: 185

v=0
o=inosmart 3945 3933 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 0 RTP/AVP 0
a=inactive
<------------->
— (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
Capabilities: us - (vp8|ulaw|alaw), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x75e057e8 – Strict RTP learning after remote address set to: 192.168.2.111:7078
Peer audio RTP is at port 192.168.2.111:7078
Peer doesn’t provide video
sip_route_dump: route/path hop: sip:inosmart@192.168.2.111:5071
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=z9hG4bK5539d92c
Max-Forwards: 70
From: “android” sip:android@192.168.2.111;tag=as27d05a17
To: sip:inosmart@192.168.2.111:5071;tag=jz-b0NY
Contact: sip:android@192.168.2.111:5060
Call-ID: 7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 13.18.3
Content-Length: 0


-- SIP/inosmart-00000035 answered SIP/android-00000034

Audio is at 17910
Video is at 192.168.2.94:17450
Adding video codec vp8 to SDP
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 192.168.2.241:8609 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.241:8609;branch=z9hG4bK-524287-1—d9b1f8603042b879;received=192.168.2.241;rport=8609
From: "android"sip:android@192.168.2.111;tag=02044f6e
To: sip:inosmart@192.168.2.111;tag=as06f67ce9
Call-ID: yJgKuzgMChtpQSE-wFFGPw…
CSeq: 2 INVITE
Server: Asterisk PBX 13.18.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:inosmart@192.168.2.94:5060
Content-Type: application/sdp
Content-Length: 358

v=0
o=root 2087320416 2087320416 IN IP4 192.168.2.94
s=Asterisk PBX 13.18.3
c=IN IP4 192.168.2.94
b=CT:384
t=0 0
m=audio 17910 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 17450 RTP/AVP 120
a=rtpmap:120 VP8/90000
a=rtcp-fb:* ccm fir
a=sendrecv

<------------>
– Channel SIP/inosmart-00000035 joined ‘simple_bridge’ basic-bridge <133ee7b3-57d9-4bdd-b31a-c24471083210>
– Channel SIP/android-00000034 joined ‘simple_bridge’ basic-bridge <133ee7b3-57d9-4bdd-b31a-c24471083210>

<— SIP read from UDP:192.168.2.241:8609 —>
ACK sip:inosmart@192.168.2.94:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.241:8609;branch=z9hG4bK-524287-1—b9c5135df9c49828;rport
Max-Forwards: 70
Contact: sip:android@192.168.2.241:8609;+sip.instance="urn:uuid:364CBC1A-AAA5-66DD-E3FF-DB5C12A21C47"
To: sip:inosmart@192.168.2.111;tag=as06f67ce9
From: "android"sip:android@192.168.2.111;tag=02044f6e
Call-ID: yJgKuzgMChtpQSE-wFFGPw…
CSeq: 2 ACK
User-Agent: PortSIP SDK for Android
Content-Length: 0

<------------->
— (10 headers 0 lines) —
> 0x75e057e8 – Strict RTP switching to RTP target address 192.168.2.111:7078 as source
> 0x75d05f90 – Strict RTP switching to RTP target address 192.168.2.241:20002 as source
> 0x75d08350 – Strict RTP switching to RTP target address 192.168.2.241:43002 as source
> 0x75d05f90 – Strict RTP learning complete - Locking on source address 192.168.2.241:20002
> 0x75e057e8 – Strict RTP learning complete - Locking on source address 192.168.2.111:7078
> 0x75d08350 – Strict RTP learning complete - Locking on source address 192.168.2.241:43002

<— SIP read from UDP:192.168.2.241:8609 —>
BYE sip:inosmart@192.168.2.94:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.2.241:8609;branch=z9hG4bK-524287-1—38fb0960d7e8ba5d;rport
Max-Forwards: 70
Contact: sip:android@192.168.2.241:8609;+sip.instance="urn:uuid:364CBC1A-AAA5-66DD-E3FF-DB5C12A21C47"
To: sip:inosmart@192.168.2.111;tag=as06f67ce9
From: "android"sip:android@192.168.2.111;tag=02044f6e
Call-ID: yJgKuzgMChtpQSE-wFFGPw…
CSeq: 3 BYE
User-Agent: PortSIP SDK for Android
Authorization: Digest username=“android”,realm=“asterisk”,nonce=“3de746c1”,uri=“sip:inosmart@192.168.2.94:5060”,response=“3e27274784e456c0309d8c4c48f523de”,algorithm=MD5
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Sending to 192.168.2.241:8609 (no NAT)
– Channel SIP/android-00000034 left ‘simple_bridge’ basic-bridge <133ee7b3-57d9-4bdd-b31a-c24471083210>
== Spawn extension (phones, inosmart, 2) exited non-zero on 'SIP/android-00000034’
Scheduling destruction of SIP dialog ‘yJgKuzgMChtpQSE-wFFGPw…’ in 38784 ms (Method: BYE)
– Channel SIP/inosmart-00000035 left ‘simple_bridge’ basic-bridge <133ee7b3-57d9-4bdd-b31a-c24471083210>

<— Transmitting (no NAT) to 192.168.2.241:8609 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.2.241:8609;branch=z9hG4bK-524287-1—38fb0960d7e8ba5d;received=192.168.2.241;rport=8609
From: "android"sip:android@192.168.2.111;tag=02044f6e
To: sip:inosmart@192.168.2.111;tag=as06f67ce9
Call-ID: yJgKuzgMChtpQSE-wFFGPw…
CSeq: 3 BYE
Server: Asterisk PBX 13.18.3
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060’ in 6400 ms (Method: INVITE)
[Dec 10 10:50:27] ERROR[745]: cdr_csv.c:315 csv_log: Unable to re-open master file /var/log/asterisk//cdr-csv//Master.csv : Permission denied
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=z9hG4bK1705c83f
Max-Forwards: 70
From: “android” sip:android@192.168.2.111;tag=as27d05a17
To: sip:inosmart@192.168.2.111:5071;tag=jz-b0NY
Call-ID: 7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 13.18.3
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


<— 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=z9hG4bK1705c83f
From: “android” sip:android@192.168.2.111;tag=as27d05a17
To: sip:inosmart@192.168.2.111:5071;tag=jz-b0NY
Call-ID: 7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060
CSeq: 103 BYE
User-Agent: (belle-sip/1.4.2)
Supported: outbound

<------------->
— (8 headers 0 lines) —
Really destroying SIP dialog ‘7682aa983755b94e7d4c8e8d7635e86d@192.168.2.111:5060’ Method: INVITE

<— SIP read from UDP:192.168.2.111:5071 —>

<------------->
Really destroying SIP dialog ‘bfwkxzQq1Rh1KvCHiQkTIg…’ Method: BYE
raspberrypiCLI> sip set debug off
SIP Debugging Disabled
[Dec 10 10:50:59] WARNING[1140]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[Dec 10 10:52:20] WARNING[1140]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[Dec 10 10:53:41] WARNING[1140]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[Dec 10 10:55:02] WARNING[1140]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[Dec 10 10:56:23] WARNING[1140]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
[Dec 10 10:57:44] WARNING[1140]: db.c:332 ast_db_put: Couldn’t execute statment: SQL logic error or missing database
raspberrypi
CLI>

The B side peer refused the video stream. Problem lies with it, rather than Asterisk.

Thank you David for you quick response.
Who is B side? softphone named Android initiated call to softphone Inosmart.
Android has camera and monitor, Inosmart has only camera. So video stream should always flow from Inosmart to Android. It does not matter which side initiated call.

A side is caller, B side is called. So B side is insomart.

insomart very clearly rejected the video stream. In particular, the a=inactive means it is definitely not prepared to send video, although specifying a port number of 0 is also sufficient to reject the video.

If it was prepared to send video, it should have provided a port number and set a=sendonly. However, I have a feeling that Asterisk would also require it to send the vp8 video codec. a=inactive means I don’t want to receive anything and I’m not going to send anything.

Thank you. I have to investigate this…

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

Looks like someone doesn’t understand direct media, or there is a firewall problem.

no, both devices are behind FW
I set directmedia=nonat,update and video stream Works in both directions :slight_smile: TNX!
Why that settings solves the problem?