Asterisk VideoCall only black Screen

Asterisk Asterisk SIP
1

Recently i came across a problem with my asterisk server. I have a Android Phone (Fanvil A32i) and a DoorPhone both registered in my Asterisk. Communication with audio only is working. But when i start to call with video there`s only a black screen showing… Therefore i called through a softphone on my Iphone (PortSip). Same here…

Dialplan is simple and in my Database

Asterisk Configuration in sip.conf:

accept_outofcall_message=yes
outofcall_message_context=message

context=from_ext                 

allowoverlap=no                 
udpbindaddr=0.0.0.0                                            
tcpenable=yes                    
tcpbindaddr=0.0.0.0                                       
transport=tcp,udp               
srvlookup=yes                   
preferred_codec_only=yes        
language=de                   

videosupport=yes            
disallow=all
allow=ulaw
allow=h264

All three sip-phones configured to ulaw and h264. Now the “strange” thing: If i look into the channels when i have an active call the log looks like this:

Owner channel ID:       SIP/SIP103-0000001a
  Our Codec Capability:   (ulaw|h264)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   (ulaw|h264)
  Joint Codec Capability:   (ulaw)
  Format:                 (ulaw|h264)

Owner channel ID:       SIP/SIP105-00000019
  Our Codec Capability:   (ulaw|h264)
  Non-Codec Capability (DTMF):   1
  Their Codec Capability:   (ulaw|gsm|alaw|g722|g729|ilbc|amr|amrwb|speex|speex16|opus|h264)
  Joint Codec Capability:   (ulaw)
  Format:                 (ulaw)

this makes absolutely no sense to me… why is the joint codec only ulaw and not h264? Maybe one of you could help. Thanks in advance

2

Maybe there is no video session. There is usually somewhere a setting for the phone that initiates video automatically. Just a guess.

3

Please provide the SDP exchanges on both legs.

4

I used tcpdump to record the networkdata. I am not that familiar with those Exchanges. Maybe u will see something. You can download the pcap here:

https://my.hidrive.com/share/bg46j.fytw

5

Too much trouble to get that into a usable form. Please see: Collecting Debug Information - Asterisk Project - Asterisk Project Wiki

6

Ok. Have done. All i really see is that there is the capability for h264 codec and the message “Welcome to Hollywood” :smiley:
New Users cant upload files so here u can download the txt…

https://my.hidrive.com/share/bg46j.fytw

7

See, in particular the section on protocol traces, Collecting Debug Information - Asterisk Project - Asterisk Project Wiki

8 
<--- SIP read from UDP:192.168.10.56:5060 --->


<------------->

<--- SIP read from UDP:192.168.10.56:5060 --->
INVITE sip:103@192.168.14.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---280a75175902c526;rport
Max-Forwards: 70
Contact: <sip:SIP105@192.168.10.56:5060>;+sip.instance="<urn:uuid:A7679F87-C916-4E88-87A4-A8532B186EE8>"
To: <sip:103@192.168.14.201>
From: <sip:SIP105@192.168.14.201>;tag=5f521304
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 1 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client iOS - v16.0.001
Allow-Events: hold, talk, conference, dialog
Content-Length: 595

v=0
o=- 1638369552 1 IN IP4 192.168.10.56
s=ps
c=IN IP4 192.168.10.56
t=0 0
m=audio 10024 RTP/AVP 105 18 8 0 3 98 99 9 101
a=rtpmap:105 opus/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:98 AMR/8000
a=rtpmap:99 AMR-WB/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=mid:audio
a=sendrecv
m=video 10028 RTP/AVP 125
a=rtpmap:125 H264/90000
a=fmtp:125 profile-level-id=42801E;packetization-mode=0
a=mid:video
a=sendrecv
a=rtcp-fb:* nack pli
<------------->
--- (14 headers 26 lines) ---
Sending to 192.168.10.56:5060 (no NAT)
Sending to 192.168.10.56:5060 (no NAT)
Using INVITE request as basis request - HUp4FScjaXpcwx3j-KULfQ..
Found peer 'SIP105' for 'SIP105' from 192.168.10.56:5060

<--- Reliably Transmitting (no NAT) to 192.168.10.56:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---280a75175902c526;received=192.168.10.56;rport=5060
From: <sip:SIP105@192.168.14.201>;tag=5f521304
To: <sip:103@192.168.14.201>;tag=as208f91c7
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 1 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="76ab396f"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'HUp4FScjaXpcwx3j-KULfQ..' in 32000 ms (Method: INVITE)

<--- SIP read from UDP:192.168.10.56:5060 --->
ACK sip:103@192.168.14.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---280a75175902c526;rport
Max-Forwards: 70
To: <sip:103@192.168.14.201>;tag=as208f91c7
From: <sip:SIP105@192.168.14.201>;tag=5f521304
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 1 ACK
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---

<--- SIP read from UDP:192.168.10.56:5060 --->
INVITE sip:103@192.168.14.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---94deca7cee23f336;rport
Max-Forwards: 70
Contact: <sip:SIP105@192.168.10.56:5060>;+sip.instance="<urn:uuid:A7679F87-C916-4E88-87A4-A8532B186EE8>"
To: <sip:103@192.168.14.201>
From: <sip:SIP105@192.168.14.201>;tag=5f521304
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 2 INVITE
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Content-Type: application/sdp
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client iOS - v16.0.001
Authorization: Digest username="SIP105",realm="asterisk",nonce="76ab396f",uri="sip:103@192.168.14.201",response="1b1530ac4b9abec35a14443b0acf1af6",algorithm=MD5
Allow-Events: hold, talk, conference, dialog
Content-Length: 595

v=0
o=- 1638369552 1 IN IP4 192.168.10.56
s=ps
c=IN IP4 192.168.10.56
t=0 0
m=audio 10024 RTP/AVP 105 18 8 0 3 98 99 9 101
a=rtpmap:105 opus/48000/2
a=fmtp:105 useinbandfec=1
a=rtpmap:18 G729/8000
a=fmtp:18 annexb=no
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:3 GSM/8000
a=rtpmap:98 AMR/8000
a=rtpmap:99 AMR-WB/16000
a=rtpmap:9 G722/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=mid:audio
a=sendrecv
m=video 10028 RTP/AVP 125
a=rtpmap:125 H264/90000
a=fmtp:125 profile-level-id=42801E;packetization-mode=0
a=mid:video
a=sendrecv
a=rtcp-fb:* nack pli
<------------->
--- (15 headers 26 lines) ---
Sending to 192.168.10.56:5060 (no NAT)
Using INVITE request as basis request - HUp4FScjaXpcwx3j-KULfQ..
Found peer 'SIP105' for 'SIP105' from 192.168.10.56:5060
Found RTP audio format 105
Found RTP audio format 18
Found RTP audio format 8
Found RTP audio format 0
Found RTP audio format 3
Found RTP audio format 98
Found RTP audio format 99
Found RTP audio format 9
Found RTP audio format 101
Found audio description format opus for ID 105
Found audio description format G729 for ID 18
Found audio description format PCMA for ID 8
Found audio description format PCMU for ID 0
Found audio description format GSM for ID 3
Found audio description format AMR for ID 98
Found audio description format AMR-WB for ID 99
Found audio description format G722 for ID 9
Found audio description format telephone-event for ID 101
Found RTP video format 125
Found video description format H264 for ID 125
Capabilities: us - (ulaw|h264), peer - audio=(ulaw|gsm|alaw|g722|g729|amr|amrwb|opus)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.56:10024
Peer video RTP is at port 192.168.10.56:10028
Looking for 103 in gonzo_sip (domain 192.168.14.201)
sip_route_dump: route/path hop: <sip:SIP105@192.168.10.56:5060>

<--- Transmitting (no NAT) to 192.168.10.56:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---94deca7cee23f336;received=192.168.10.56;rport=5060
From: <sip:SIP105@192.168.14.201>;tag=5f521304
To: <sip:103@192.168.14.201>
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:103@192.168.14.201:5060>
Content-Length: 0


<------------>
Audio is at 13550
Adding codec ulaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<--- Reliably Transmitting (no NAT) to 192.168.10.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---94deca7cee23f336;received=192.168.10.56;rport=5060
From: <sip:SIP105@192.168.14.201>;tag=5f521304
To: <sip:103@192.168.14.201>;tag=as18e4b1e8
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 2 INVITE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:103@192.168.14.201:5060>
Content-Type: application/sdp
Content-Length: 279

v=0
o=root 571180160 571180160 IN IP4 192.168.14.201
s=Asterisk PBX 16.2.1~dfsg-1+deb10u2
c=IN IP4 192.168.14.201
t=0 0
m=audio 13550 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 0 RTP/AVP 125

<------------>

<--- SIP read from UDP:192.168.10.56:5060 --->
ACK sip:103@192.168.14.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---b227410078e1ff3c;rport
Max-Forwards: 70
Contact: <sip:SIP105@192.168.10.56:5060>;+sip.instance="<urn:uuid:A7679F87-C916-4E88-87A4-A8532B186EE8>"
To: <sip:103@192.168.14.201>;tag=as18e4b1e8
From: <sip:SIP105@192.168.14.201>;tag=5f521304
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 2 ACK
User-Agent: PortSIP UC Client iOS - v16.0.001
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Audio is at 19536
Video is at 192.168.14.201:18624
Adding codec ulaw to SDP
Adding video codec h264 to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.168.10.60:5060:
INVITE sip:SIP103@192.168.10.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK1d714071
Max-Forwards: 70
From: <sip:105@192.168.14.201>;tag=as1e078ee9
To: <sip:SIP103@192.168.10.60:5060>
Contact: <sip:105@192.168.14.201:5060>
Call-ID: 1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Date: Wed, 01 Dec 2021 13:17:32 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 330

v=0
o=root 972994088 972994088 IN IP4 192.168.14.201
s=Asterisk PBX 16.2.1~dfsg-1+deb10u2
c=IN IP4 192.168.14.201
b=CT:384
t=0 0
m=audio 19536 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv
m=video 18624 RTP/AVP 125
a=rtpmap:125 H264/90000
a=sendrecv

---

<--- SIP read from UDP:192.168.10.60:5060 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK1d714071
From: <sip:105@192.168.14.201>;tag=as1e078ee9
To: <sip:SIP103@192.168.10.60:5060>
Call-ID: 1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060
CSeq: 102 INVITE
User-Agent: Fanvil A32i 2.6.0.408 0c383e16941e
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---

<--- SIP read from UDP:192.168.10.60:5060 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK1d714071
From: <sip:105@192.168.14.201>;tag=as1e078ee9
To: <sip:SIP103@192.168.10.60:5060>;tag=665152082
Call-ID: 1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060
CSeq: 102 INVITE
Contact: <sip:SIP103@192.168.10.60:5060>
User-Agent: Fanvil A32i 2.6.0.408 0c383e16941e
Allow-Events: talk,hold
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:SIP103@192.168.10.60:5060>

<--- SIP read from UDP:192.168.10.69:5504 --->
OPTIONS sip:192.168.14.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.69:5504;branch=z9hG4bK29207304952989219472;rport
From: SIP104 <sip:SIP104@192.168.14.201:5060>;tag=641018359
To: <sip:192.168.14.201:5060>
Call-ID: 48761537532346-291672156218891@192.168.10.69
CSeq: 1 OPTIONS
Max-Forwards: 70
User-Agent: Fanvil i32V 2.8.0.6949 0c383e1575da
Accept: application/sdp
Content-Length: 0

<------------->
--- (10 headers 0 lines) ---
Sending to 192.168.10.69:5504 (no NAT)
Looking for s in from_ext (domain 192.168.14.201)

<--- Transmitting (no NAT) to 192.168.10.69:5504 --->
SIP/2.0 404 Not Found
Via: SIP/2.0/UDP 192.168.10.69:5504;branch=z9hG4bK29207304952989219472;received=192.168.10.69;rport=5504
From: SIP104 <sip:SIP104@192.168.14.201:5060>;tag=641018359
To: <sip:192.168.14.201:5060>;tag=as5af87eb4
Call-ID: 48761537532346-291672156218891@192.168.10.69
CSeq: 1 OPTIONS
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Accept: application/sdp
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '48761537532346-291672156218891@192.168.10.69' in 32000 ms (Method: OPTIONS)

<--- SIP read from UDP:192.168.10.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK1d714071
From: <sip:105@192.168.14.201>;tag=as1e078ee9
To: <sip:SIP103@192.168.10.60:5060>;tag=665152082
Call-ID: 1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060
CSeq: 102 INVITE
Contact: <sip:SIP103@192.168.10.60:5060>
Supported: 100rel, replaces, timer
User-Agent: Fanvil A32i 2.6.0.408 0c383e16941e
Allow-Events: talk,hold
Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REFER, NOTIFY, INFO, PRACK, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length: 362

v=0
o=SIP103 1445252969 448639290 IN IP4 192.168.10.60
s=A conversation
c=IN IP4 192.168.10.60
t=0 0
m=audio 10132 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:150
a=sendrecv
m=video 10134 RTP/AVP 125
a=rtpmap:125 H264/90000
a=fmtp:125 profile-level-id=42801f; max-br=1000
a=sendrecv
<------------->
--- (13 headers 16 lines) ---
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Found RTP video format 125
Found video description format H264 for ID 125
Capabilities: us - (ulaw|h264), peer - audio=(ulaw)/video=(h264)/text=(nothing), combined - (ulaw|h264)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 192.168.10.60:10132
Peer video RTP is at port 192.168.10.60:10134
sip_route_dump: route/path hop: <sip:SIP103@192.168.10.60:5060>
set_destination: Parsing <sip:SIP103@192.168.10.60:5060> for address/port to send to
set_destination: set destination to 192.168.10.60:5060
Transmitting (no NAT) to 192.168.10.60:5060:
ACK sip:SIP103@192.168.10.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK56b623b9
Max-Forwards: 70
From: <sip:105@192.168.14.201>;tag=as1e078ee9
To: <sip:SIP103@192.168.10.60:5060>;tag=665152082
Contact: <sip:105@192.168.14.201:5060>
Call-ID: 1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Content-Length: 0


---

<--- SIP read from UDP:192.168.10.60:5060 --->

<------------->

<--- SIP read from UDP:192.168.10.56:5060 --->


<------------->

<--- SIP read from UDP:192.168.10.56:5060 --->
BYE sip:103@192.168.14.201:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---28ae8d19f9bdb01d;rport
Max-Forwards: 70
Contact: <sip:SIP105@192.168.10.56:5060>;+sip.instance="<urn:uuid:A7679F87-C916-4E88-87A4-A8532B186EE8>"
To: <sip:103@192.168.14.201>;tag=as18e4b1e8
From: <sip:SIP105@192.168.14.201>;tag=5f521304
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 3 BYE
User-Agent: PortSIP UC Client iOS - v16.0.001
Authorization: Digest username="SIP105",realm="asterisk",nonce="76ab396f",uri="sip:103@192.168.14.201:5060",response="0b01c5462543546b40748986fb95558d",algorithm=MD5
Content-Length: 0

<------------->
--- (11 headers 0 lines) ---
Sending to 192.168.10.56:5060 (no NAT)
Scheduling destruction of SIP dialog 'HUp4FScjaXpcwx3j-KULfQ..' in 32000 ms (Method: BYE)

<--- Transmitting (no NAT) to 192.168.10.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---28ae8d19f9bdb01d;received=192.168.10.56;rport=5060
From: <sip:SIP105@192.168.14.201>;tag=5f521304
To: <sip:103@192.168.14.201>;tag=as18e4b1e8
Call-ID: HUp4FScjaXpcwx3j-KULfQ..
CSeq: 3 BYE
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog '1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060' in 32000 ms (Method: INVITE)
set_destination: Parsing <sip:SIP103@192.168.10.60:5060> for address/port to send to
set_destination: set destination to 192.168.10.60:5060
Reliably Transmitting (no NAT) to 192.168.10.60:5060:
BYE sip:SIP103@192.168.10.60:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK3a79b7a7
Max-Forwards: 70
From: <sip:105@192.168.14.201>;tag=as1e078ee9
To: <sip:SIP103@192.168.10.60:5060>;tag=665152082
Call-ID: 1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060
CSeq: 103 BYE
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0


---

<--- SIP read from UDP:192.168.10.60:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK3a79b7a7
From: <sip:105@192.168.14.201>;tag=as1e078ee9
To: <sip:SIP103@192.168.10.60:5060>;tag=665152082
Call-ID: 1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060
CSeq: 103 BYE
User-Agent: Fanvil A32i 2.6.0.408 0c383e16941e
Content-Length: 0

<------------->
--- (8 headers 0 lines) ---
Really destroying SIP dialog '1dc9fd7e0c4a834716a7fb086cf5c77f@192.168.14.201:5060' Method: INVITE

<--- SIP read from UDP:192.168.10.56:5060 --->


<------------->

<--- SIP read from UDP:192.168.10.56:5060 --->
REGISTER sip:192.168.14.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---6ce4dc4180730e44;rport
Max-Forwards: 70
Contact: <sip:SIP105@192.168.10.56:5060>;+sip.instance="<urn:uuid:A7679F87-C916-4E88-87A4-A8532B186EE8>"
To: <sip:SIP105@192.168.14.201>
From: <sip:SIP105@192.168.14.201>;tag=3676097d
Call-ID: OOld7Sia7xJpaAu6kPgQKg..
CSeq: 109 REGISTER
Expires: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client iOS - v16.0.001
Authorization: Digest username="SIP105",realm="asterisk",nonce="4b6304bd",uri="sip:192.168.14.201",response="4f8e4a0c64241956f9b0654e0a8fcc7c",algorithm=MD5
Allow-Events: hold, talk, conference, dialog
x-p-push: device-os=ios;device-uid=e07e3997e2cf5aba38d2a23fa34ec7132a1aeedd2439fef73151936ff51e1ef6|d87e28381f44c19ed45d35dc2c8efebc96b771b03812c5f80dd2eb7e37482e3a;allow-call-push=true;allow-message-push=true;app-id=com.portsip.portgo
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.10.56:5060 (no NAT)
Sending to 192.168.10.56:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.10.56:5060 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---6ce4dc4180730e44;received=192.168.10.56;rport=5060
From: <sip:SIP105@192.168.14.201>;tag=3676097d
To: <sip:SIP105@192.168.14.201>;tag=as79518040
Call-ID: OOld7Sia7xJpaAu6kPgQKg..
CSeq: 109 REGISTER
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="1cfff219"
Content-Length: 0


<------------>
Scheduling destruction of SIP dialog 'OOld7Sia7xJpaAu6kPgQKg..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.10.56:5060 --->
REGISTER sip:192.168.14.201 SIP/2.0
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---2307703610f09d3d;rport
Max-Forwards: 70
Contact: <sip:SIP105@192.168.10.56:5060>;+sip.instance="<urn:uuid:A7679F87-C916-4E88-87A4-A8532B186EE8>"
To: <sip:SIP105@192.168.14.201>
From: <sip:SIP105@192.168.14.201>;tag=3676097d
Call-ID: OOld7Sia7xJpaAu6kPgQKg..
CSeq: 110 REGISTER
Expires: 90
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, REGISTER, SUBSCRIBE, INFO, PUBLISH
Supported: replaces, answermode, eventlist, outbound, path
User-Agent: PortSIP UC Client iOS - v16.0.001
Authorization: Digest username="SIP105",realm="asterisk",nonce="1cfff219",uri="sip:192.168.14.201",response="d929bfaaef6459bb59e1e98af46fb6d2",algorithm=MD5
Allow-Events: hold, talk, conference, dialog
x-p-push: device-os=ios;device-uid=e07e3997e2cf5aba38d2a23fa34ec7132a1aeedd2439fef73151936ff51e1ef6|d87e28381f44c19ed45d35dc2c8efebc96b771b03812c5f80dd2eb7e37482e3a;allow-call-push=true;allow-message-push=true;app-id=com.portsip.portgo
Content-Length: 0

<------------->
--- (16 headers 0 lines) ---
Sending to 192.168.10.56:5060 (no NAT)

<--- Transmitting (no NAT) to 192.168.10.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.10.56:5060;branch=z9hG4bK-524287-1---2307703610f09d3d;received=192.168.10.56;rport=5060
From: <sip:SIP105@192.168.14.201>;tag=3676097d
To: <sip:SIP105@192.168.14.201>;tag=as79518040
Call-ID: OOld7Sia7xJpaAu6kPgQKg..
CSeq: 110 REGISTER
Server: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Expires: 90
Contact: <sip:SIP105@192.168.10.56:5060>;expires=90
Date: Wed, 01 Dec 2021 13:18:02 GMT
Content-Length: 0


<------------>
Reliably Transmitting (no NAT) to 192.168.10.56:5060:
NOTIFY sip:SIP105@192.168.10.56:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK71281205;rport
Max-Forwards: 70
Route: <sip:SIP105@192.168.10.56:5060>
From: "asterisk" <sip:asterisk@192.168.14.201>;tag=as67e94d5e
To: <sip:SIP105@192.168.10.56:5060>;tag=d9377c28
Contact: <sip:asterisk@192.168.14.201:5060>
Call-ID: OSImEUmaTRK48WBLTcdwbQ..
CSeq: 156 NOTIFY
User-Agent: Asterisk PBX 16.2.1~dfsg-1+deb10u2
Event: message-summary
Content-Type: application/simple-message-summary
Subscription-State: active
Content-Length: 94

Messages-Waiting: no
Message-Account: sip:asterisk@192.168.14.201
Voice-Message: 0/0 (0/0)

---
Scheduling destruction of SIP dialog 'OOld7Sia7xJpaAu6kPgQKg..' in 32000 ms (Method: REGISTER)

<--- SIP read from UDP:192.168.10.56:5060 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.14.201:5060;branch=z9hG4bK71281205;rport=5060
Contact: <sip:SIP105@192.168.10.56:5060>;+sip.instance="<urn:uuid:A7679F87-C916-4E88-87A4-A8532B186EE8>"
To: <sip:SIP105@192.168.10.56:5060>;tag=d9377c28
From: "asterisk" <sip:asterisk@192.168.14.201>;tag=as67e94d5e
Call-ID: OSImEUmaTRK48WBLTcdwbQ..
CSeq: 156 NOTIFY
User-Agent: PortSIP UC Client iOS - v16.0.001
Content-Length: 0

<------------->
--- (9 headers 0 lines) ---
9

It has rejected the A side video stream before even touching the B side. I can’t see why.

However 16.2.1 is very old and will be buggy, so you should retry with 16.22.0, or take up the issue with the Debian packager.

10

THanks for your help. I will try it out :slight_smile:

11

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