Setting up Asterisk and webrtc

Can anyone help me configure Asterisk to work simultaneously with local video intercom panel “Bas IP” and clients connecting via webrtc? When calling from a browser there is a connection, sound in both directions, but no video.

Nginx has SSL certificates installed and clients connect without problems

I use last version of Asterisk (20.6.0), use Nginx to reverse proxy, and Kerio VPN client to connect the server with video intercom panels into one VPN network.

Maybe there is something in the configuration files that I missed?

Here are my config files:

**pjsip.conf**

[global]
max_forwards=70
keep_alive_interval=300

; == Transports

[udp_transport]
type=transport
protocol=udp
bind=0.0.0.0
tos=af42
cos=3

[wss_transport]
type=transport
protocol=wss
bind=0.0.0.0


; == ACL

[acl] ; Opperates on all pjsip traffic (can also be in acl.conf)
type=acl
;deny=0.0.0.0/0.0.0.0
;permit=10.0.0.0/255.0.0.0
;permit=172.16.0.0/255.240.0.0
;permit=192.168.0.0/255.255.0.0
;permit=127.0.0.1/255.255.255.255
;permit=192.168.58.0/255.255.255.0
permit=0.0.0.0/0.0.0.0

; == Templates

[single_aor](!)
max_contacts=1
qualify_frequency=120
remove_existing=yes

[userpass_auth](!)
auth_type=userpass

[basic_endpoint](!)
moh_suggest=default
context=from-extensions
inband_progress=no
rtp_timeout=120
message_context=textmessages
allow_subscribe=yes
subscribe_context=subscriptions
direct_media=no
dtmf_mode=rfc4733
device_state_busy_at=1
disallow=all

[phone_endpoint](!)
;allow=ulaw,alaw,g722,gsm,vp9,vp8,h264
allow=alaw,ulaw,h264

[webrtc_endpoint](!)
transport=wss_transport
;allow=opus,ulaw,vp9,vp8,h264
allow=alaw,ulaw,h264
webrtc=yes
dtls_auto_generate_cert=yes
max_audio_streams=5
max_video_streams=16
media_use_received_transport=yes
rtp_symmetric=yes
rewrite_contact=yes
force_rport=yes

; == Users

[User1](basic_endpoint,webrtc_endpoint)
type=endpoint
callerid="One Hundred" <100>
auth=User1
aors=User1
[User1](single_aor)
type=aor
mailboxes=User1@default
[User1](userpass_auth)
type=auth
username=User1
password=some_strong_password

[User2](basic_endpoint,webrtc_endpoint)
type=endpoint
callerid="Two Hundred" <200>
auth=User2
aors=User2
[User2](single_aor)
type=aor
[User2](userpass_auth)
type=auth
username=User2
password=some_strong_password

[User3](basic_endpoint,phone_endpoint)
type=endpoint
callerid="Three Hundred" <300>
auth=User3
aors=User3
[User3](single_aor)
type=aor
[User3](userpass_auth)
type=auth
username=User3
password=some_strong_password

[User4](basic_endpoint,phone_endpoint)
type=endpoint
callerid="Four Hundred" <400>
auth=User4
aors=User4
[User4](single_aor)
type=aor
[User4](userpass_auth)
type=auth
username=User4
password=some_strong_password
**http.conf**

[general]
enabled=yes ; HTTP 
bindaddr=127.0.0.1 
bindport=8080 
tlsenable=no ; HTTPS 
enablestatic=no 

**extensions.conf**

[general]
static=yes
writeprotect=yes
priorityjumping=no
autofallthrough=no

[globals]

[subscriptions]
exten => 100,hint,PJSIP/User1
exten => 200,hint,PJSIP/User2
exten => 300,hint,PJSIP/User3
exten => 400,hint,PJSIP/User4


[from-extensions]
exten => 100,1,Dial(PJSIP/User1,30)
exten => 200,1,Dial(PJSIP/User2,30)
exten => 300,1,Dial(PJSIP/User3,30)
exten => 400,1,Dial(PJSIP/User4,30)

exten => _[*0-9].,1,NoOp(Music On Hold)
exten => _[*0-9].,n,Ringing()
exten => _[*0-9].,n,Wait(2)
exten => _[*0-9].,n,Answer()
exten => _[*0-9].,n,Wait(1)
exten => _[*0-9].,n,MusicOnHold()

exten => e,1,Hangup()

What do the INVITE and its response (the SIP packets containing the SDP)
contain?

Antony.

Here’s the dump file from sngrep:

2024/02/12 16:22:05.922088 127.0.0.1:49088 -> 127.0.0.1:8080
REGISTER sip:sip.myserv.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.128;branch=z9hG4bK8929873
To: <sip:User1@sip.myserv.com>
From: "Phone" <sip:User1@sip.myserv.com>;tag=9ijl3tbcfd
CSeq: 12 REGISTER
Call-ID: c0m4pr6upp0crsne5ort
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="User1", realm="sip.myserv.com", nonce="1707754447/b849d97a2808c8eb705b4edc694ed9d8", uri="sip:sip.myserv.com", response="7dce855bed88ccad63bca464e47bc186", opaque="1c5dc7621db765a9", qop=auth, cnonce="cvg6nfi7jde2", nc=00000001
Contact: <sip:haml1iie@192.0.2.128;transport=wss>;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.24 (SIPJS - 0.20.0) Mozilla/5.0 (Linux; Android 10; K) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/121.0.0.0 Mobile Safari/537.36
Content-Length: 0



2024/02/12 16:22:05.922511 127.0.0.1:8080 -> 127.0.0.1:49088
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.128;rport=49088;received=127.0.0.1;branch=z9hG4bK8929873
Call-ID: c0m4pr6upp0crsne5ort
From: "Phone" <sip:User1@sip.myserv.com>;tag=9ijl3tbcfd
To: <sip:User1@sip.myserv.com>;tag=z9hG4bK8929873
CSeq: 12 REGISTER
WWW-Authenticate: Digest realm="sip.myserv.com",nonce="1707754925/245842e6cf39976a8d5151bbce75767b",opaque="2d5ed2f94ed749f8",stale=true,algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.6.0
Content-Length:  0



2024/02/12 16:22:05.953262 127.0.0.1:49088 -> 127.0.0.1:8080
REGISTER sip:sip.myserv.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.128;branch=z9hG4bK5985234
To: <sip:User1@sip.myserv.com>
From: "Phone" <sip:User1@sip.myserv.com>;tag=9ijl3tbcfd
CSeq: 13 REGISTER
Call-ID: c0m4pr6upp0crsne5ort
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="User1", realm="sip.myserv.com", nonce="1707754925/245842e6cf39976a8d5151bbce75767b", uri="sip:sip.myserv.com", response="6487d841aad817830623b64f7928b61a", opaque="2d5ed2f94ed749f8", qop=auth, cnonce="3in6hjvkokvt", nc=00000001
Contact: <sip:haml1iie@192.0.2.128;transport=wss>;expires=300
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound, path, gruu
User-Agent: Browser Phone 0.3.24 (SIPJS - 0.20.0) Mozilla/5.0 (Linux; Android 10; K) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/121.0.0.0 Mobile Safari/537.36
Content-Length: 0



2024/02/12 16:22:05.954258 127.0.0.1:8080 -> 127.0.0.1:49088
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.128;rport=49088;received=127.0.0.1;branch=z9hG4bK5985234
Call-ID: c0m4pr6upp0crsne5ort
From: "Phone" <sip:User1@sip.myserv.com>;tag=9ijl3tbcfd
To: <sip:User1@sip.myserv.com>;tag=z9hG4bK5985234
CSeq: 13 REGISTER
Date: Mon, 12 Feb 2024 16:22:05 GMT
Contact: <sip:haml1iie@192.0.2.128;transport=WS>;expires=299
Server: Asterisk PBX 20.6.0
Content-Length:  0



2024/02/12 16:22:05.954660 127.0.0.1:8080 -> 127.0.0.1:49088
OPTIONS sip:haml1iie@127.0.0.1:49088;transport=WS SIP/2.0
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPjb037824b-f68c-491d-afb9-e013630f2c9a;alias
From: <sip:User1@sip.myserv.com>;tag=a042b2ae-50b9-43a3-9ca5-3a4973b67a07
To: <sip:haml1iie@127.0.0.1>
Contact: <sip:User1@sip.myserv.com:5060;transport=ws>
Call-ID: 8c350b57-204f-42a9-9c5d-768972d3ba3f
CSeq: 7497 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Length:  0



2024/02/12 16:22:05.988891 127.0.0.1:49088 -> 127.0.0.1:8080
SIP/2.0 200 OK
Via: SIP/2.0/WS 127.0.0.1:8080;rport;branch=z9hG4bKPjb037824b-f68c-491d-afb9-e013630f2c9a;alias
From: <sip:User1@sip.myserv.com>;tag=a042b2ae-50b9-43a3-9ca5-3a4973b67a07
To: <sip:haml1iie@127.0.0.1>;tag=ablsrmbrt5
CSeq: 7497 OPTIONS
Call-ID: 8c350b57-204f-42a9-9c5d-768972d3ba3f
Supported: outbound
User-Agent: Browser Phone 0.3.24 (SIPJS - 0.20.0) Mozilla/5.0 (Linux; Android 10; K) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/121.0.0.0 Mobile Safari/537.36
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Accept: application/sdp,application/dtmf-relay
Content-Length: 0



2024/02/12 16:22:12.312884 127.0.0.1:49088 -> 127.0.0.1:8080
INVITE sip:300@sip.myserv.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.128;branch=z9hG4bK5418602
To: <sip:300@sip.myserv.com>
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
CSeq: 1 INVITE
Call-ID: c0m4pdnp0q083l8hbsgu
Max-Forwards: 70
Contact: <sip:haml1iie@192.0.2.128;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.24 (SIPJS - 0.20.0) Mozilla/5.0 (Linux; Android 10; K) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/121.0.0.0 Mobile Safari/537.36
Content-Type: application/sdp
Content-Length: 5527

v=0
o=- 4274014433033588599 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS feb2c208-dcad-43d4-94ed-4460c0e8af3f
m=audio 52904 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 my_local_pc_ip
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1346443971 1 udp 2122260223 192.168.1.104 52904 typ host generation 0 network-id 2 network-cost 10
a=candidate:2303628187 1 udp 2122194687 26.194.2.74 40942 typ host generation 0 network-id 1 network-cost 900
a=candidate:2186125132 1 udp 1686052607 my_local_pc_ip 52904 typ srflx raddr 192.168.1.104 rport 52904 generation 0 network-id 2 network-cost 10
a=candidate:2934689367 1 tcp 1518280447 192.168.1.104 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:2011449103 1 tcp 1518214911 26.194.2.74 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:YULh
a=ice-pwd:s7roM6yJ7AAUy3VFsTREKkJP
a=ice-options:trickle
a=fingerprint:sha-256 9C:24:FC:3D:98:0A:C2:E1:24:A5:EE:24:32:49:15:49:27:E1:EB:B4:19:22:1D:60:88:38:41:82:7B:45:84:DA
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:feb2c208-dcad-43d4-94ed-4460c0e8af3f 7afef39c-ea55-40d4-91f5-c7e41024f7e8
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2768622923 cname:UJDdCt0SAmXUh3e3
a=ssrc:2768622923 msid:feb2c208-dcad-43d4-94ed-4460c0e8af3f 7afef39c-ea55-40d4-91f5-c7e41024f7e8
m=video 44552 UDP/TLS/RTP/SAVPF 96 97 98 99 112 113 106 107 108 109 100 101 116 117 118
c=IN IP4 my_local_pc_ip
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1346443971 1 udp 2122260223 192.168.1.104 44552 typ host generation 0 network-id 2 network-cost 10
a=candidate:2303628187 1 udp 2122194687 26.194.2.74 58483 typ host generation 0 network-id 1 network-cost 900
a=candidate:2186125132 1 udp 1686052607 my_local_pc_ip 44552 typ srflx raddr 192.168.1.104 rport 44552 generation 0 network-id 2 network-cost 10
a=candidate:2934689367 1 tcp 1518280447 192.168.1.104 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:2011449103 1 tcp 1518214911 26.194.2.74 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:YULh
a=ice-pwd:s7roM6yJ7AAUy3VFsTREKkJP
a=ice-options:trickle
a=fingerprint:sha-256 9C:24:FC:3D:98:0A:C2:E1:24:A5:EE:24:32:49:15:49:27:E1:EB:B4:19:22:1D:60:88:38:41:82:7B:45:84:DA
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/playout-delay
a=extmap:6 http://www.webrtc.org/experiments/rtp-hdrext/video-content-type
a=extmap:7 http://www.webrtc.org/experiments/rtp-hdrext/video-timing
a=extmap:8 http://www.webrtc.org/experiments/rtp-hdrext/color-space
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=extmap:10 u

2024/02/12 16:22:12.313246 127.0.0.1:8080 -> 127.0.0.1:49088
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.128;rport=49088;received=127.0.0.1;branch=z9hG4bK5418602
Call-ID: c0m4pdnp0q083l8hbsgu
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
To: <sip:300@sip.myserv.com>;tag=z9hG4bK5418602
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="sip.myserv.com",nonce="1707754932/89fedf021baa75f6e9a1498db287137d",opaque="55f36b1e393fd244",algorithm=MD5,qop="auth"
Server: Asterisk PBX 20.6.0
Content-Length:  0



2024/02/12 16:22:12.353784 127.0.0.1:49088 -> 127.0.0.1:8080
ACK sip:300@sip.myserv.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.128;branch=z9hG4bK5418602
To: <sip:300@sip.myserv.com>;tag=z9hG4bK5418602
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
Call-ID: c0m4pdnp0q083l8hbsgu
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0



2024/02/12 16:22:12.355703 127.0.0.1:49088 -> 127.0.0.1:8080
INVITE sip:300@sip.myserv.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.128;branch=z9hG4bK6071678
To: <sip:300@sip.myserv.com>
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
CSeq: 2 INVITE
Call-ID: c0m4pdnp0q083l8hbsgu
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="User1", realm="sip.myserv.com", nonce="1707754932/89fedf021baa75f6e9a1498db287137d", uri="sip:300@sip.myserv.com", response="13152c67714009ee7e0e4e60ef45311f", opaque="55f36b1e393fd244", qop=auth, cnonce="0tptvhmsu1q1", nc=00000001
Contact: <sip:haml1iie@192.0.2.128;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.24 (SIPJS - 0.20.0) Mozilla/5.0 (Linux; Android 10; K) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/121.0.0.0 Mobile Safari/537.36
Content-Type: application/sdp
Content-Length: 5527

v=0
o=- 4274014433033588599 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0 1
a=extmap-allow-mixed
a=msid-semantic: WMS feb2c208-dcad-43d4-94ed-4460c0e8af3f
m=audio 52904 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 my_local_pc_ip
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1346443971 1 udp 2122260223 192.168.1.104 52904 typ host generation 0 network-id 2 network-cost 10
a=candidate:2303628187 1 udp 2122194687 26.194.2.74 40942 typ host generation 0 network-id 1 network-cost 900
a=candidate:2186125132 1 udp 1686052607 my_local_pc_ip 52904 typ srflx raddr 192.168.1.104 rport 52904 generation 0 network-id 2 network-cost 10
a=candidate:2934689367 1 tcp 1518280447 192.168.1.104 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:2011449103 1 tcp 1518214911 26.194.2.74 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:YULh
a=ice-pwd:s7roM6yJ7AAUy3VFsTREKkJP
a=ice-options:trickle
a=fingerprint:sha-256 9C:24:FC:3D:98:0A:C2:E1:24:A5:EE:24:32:49:15:49:27:E1:EB:B4:19:22:1D:60:88:38:41:82:7B:45:84:DA
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid
a=sendrecv
a=msid:feb2c208-dcad-43d4-94ed-4460c0e8af3f 7afef39c-ea55-40d4-91f5-c7e41024f7e8
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 minptime=10;useinbandfec=1
a=rtpmap:63 red/48000/2
a=fmtp:63 111/111
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:126 telephone-event/8000
a=ssrc:2768622923 cname:UJDdCt0SAmXUh3e3
a=ssrc:2768622923 msid:feb2c208-dcad-43d4-94ed-4460c0e8af3f 7afef39c-ea55-40d4-91f5-c7e41024f7e8
m=video 44552 UDP/TLS/RTP/SAVPF 96 97 98 99 112 113 106 107 108 109 100 101 116 117 118
c=IN IP4 my_local_pc_ip
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1346443971 1 udp 2122260223 192.168.1.104 44552 typ host generation 0 network-id 2 network-cost 10
a=candidate:2303628187 1 udp 2122194687 26.194.2.74 58483 typ host generation 0 network-id 1 network-cost 900
a=candidate:2186125132 1 udp 1686052607 my_local_pc_ip 44552 typ srflx raddr 192.168.1.104 rport 44552 generation 0 network-id 2 network-cost 10
a=candidate:2934689367 1 tcp 1518280447 192.168.1.104 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:2011449103 1 tcp 1518214911 26.194.2.74 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=ice-ufrag:YULh
a=ice-pwd:s7roM6yJ7AAUy3VFsTREKkJP
a=ice-options:trickle
a=fingerprint:sha-256 9C:24:FC:3D:98:0A:C2:E1:24:A5:EE:24:32:49:15:49:27:E1:EB:B4:19:22:1D:60:88:38:41:82:7B:45:84:DA
a=setup:actpass
a=mid:1
a=extmap:14 urn:ietf:params:rtp-hdrext:toffset
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:13 urn:3gpp:video-orientation
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=extmap:5 http://www.webrtc.org/experiments/rtp-hdrext/p

2024/02/12 16:22:12.356415 127.0.0.1:8080 -> 127.0.0.1:49088
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.128;rport=49088;received=127.0.0.1;branch=z9hG4bK6071678
Call-ID: c0m4pdnp0q083l8hbsgu
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
To: <sip:300@sip.myserv.com>
CSeq: 2 INVITE
Server: Asterisk PBX 20.6.0
Content-Length:  0



2024/02/12 16:22:12.422302 127.0.0.1:8080 -> 127.0.0.1:49088
SIP/2.0 183 Session Progress
Via: SIP/2.0/WSS 192.0.2.128;rport=49088;received=127.0.0.1;branch=z9hG4bK6071678
Call-ID: c0m4pdnp0q083l8hbsgu
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
To: <sip:300@sip.myserv.com>;tag=c2c63d2d-76bf-4fc9-8394-d2807163001e
CSeq: 2 INVITE
Server: Asterisk PBX 20.6.0
Contact: <sip:127.0.0.1:8080;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Content-Type: application/sdp
Content-Length:  1685

v=0
o=- 1843138423 4 IN IP4 my_server_ip
s=Asterisk
c=IN IP4 my_server_ip
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0 1
m=audio 16628 UDP/TLS/RTP/SAVPF 8 0 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 E8:8B:FD:42:DD:C9:EE:69:F1:23:DB:DA:19:EA:3E:2C:8F:93:A6:47:D0:9D:1B:A3:BA:99:F5:8C:B7:B0:1D:C6
a=ice-ufrag:68276ae54eb2f48425b286a47ac2ee06
a=ice-pwd:1355b9bf49313a396bc546c2568bc08e
a=candidate:Hbc484d0e 1 UDP 2130706431 my_server_ip 16628 typ host
a=candidate:Hc0a83a06 1 UDP 2130706431 192.168.58.6 16628 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:1656293961 cname:64aa6180-7ce0-4309-85b4-8be3408f971c
a=msid:71f6c8ca-29a9-4896-a09a-dda51f36b9c1 efe90d80-a1f7-4734-bec3-336db8f32a9a
a=rtcp-fb:* transport-cc
a=mid:0
m=video 16628 UDP/TLS/RTP/SAVPF 112
a=connection:new
a=setup:active
a=fingerprint:SHA-256 E8:8B:FD:42:DD:C9:EE:69:F1:23:DB:DA:19:EA:3E:2C:8F:93:A6:47:D0:9D:1B:A3:BA:99:F5:8C:B7:B0:1D:C6
a=ice-ufrag:68276ae54eb2f48425b286a47ac2ee06
a=ice-pwd:1355b9bf49313a396bc546c2568bc08e
a=rtpmap:112 H264/90000
a=fmtp:112 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42E01F
a=sendrecv
a=rtcp-mux
a=ssrc:1818874216 cname:d7ff75df-cbc6-4797-aa59-0a694eb42e90
a=msid:71f6c8ca-29a9-4896-a09a-dda51f36b9c1 5e6b108c-6aa7-4ba6-9b26-d4a595b6d4b7
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtcp-fb:* nack
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=mid:1


2024/02/12 16:22:13.296208 127.0.0.1:8080 -> 127.0.0.1:49088
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.128;rport=49088;received=127.0.0.1;branch=z9hG4bK6071678
Call-ID: c0m4pdnp0q083l8hbsgu
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
To: <sip:300@sip.myserv.com>;tag=c2c63d2d-76bf-4fc9-8394-d2807163001e
CSeq: 2 INVITE
Server: Asterisk PBX 20.6.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Contact: <sip:127.0.0.1:8080;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:  1685

v=0
o=- 1843138423 4 IN IP4 my_server_ip
s=Asterisk
c=IN IP4 my_server_ip
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0 1
m=audio 16628 UDP/TLS/RTP/SAVPF 8 0 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 E8:8B:FD:42:DD:C9:EE:69:F1:23:DB:DA:19:EA:3E:2C:8F:93:A6:47:D0:9D:1B:A3:BA:99:F5:8C:B7:B0:1D:C6
a=ice-ufrag:68276ae54eb2f48425b286a47ac2ee06
a=ice-pwd:1355b9bf49313a396bc546c2568bc08e
a=candidate:Hbc484d0e 1 UDP 2130706431 my_server_ip 16628 typ host
a=candidate:Hc0a83a06 1 UDP 2130706431 192.168.58.6 16628 typ host
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
a=rtcp-mux
a=ssrc:1656293961 cname:64aa6180-7ce0-4309-85b4-8be3408f971c
a=msid:71f6c8ca-29a9-4896-a09a-dda51f36b9c1 efe90d80-a1f7-4734-bec3-336db8f32a9a
a=rtcp-fb:* transport-cc
a=mid:0
m=video 16628 UDP/TLS/RTP/SAVPF 112
a=connection:new
a=setup:active
a=fingerprint:SHA-256 E8:8B:FD:42:DD:C9:EE:69:F1:23:DB:DA:19:EA:3E:2C:8F:93:A6:47:D0:9D:1B:A3:BA:99:F5:8C:B7:B0:1D:C6
a=ice-ufrag:68276ae54eb2f48425b286a47ac2ee06
a=ice-pwd:1355b9bf49313a396bc546c2568bc08e
a=rtpmap:112 H264/90000
a=fmtp:112 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42E01F
a=sendrecv
a=rtcp-mux
a=ssrc:1818874216 cname:d7ff75df-cbc6-4797-aa59-0a694eb42e90
a=msid:71f6c8ca-29a9-4896-a09a-dda51f36b9c1 5e6b108c-6aa7-4ba6-9b26-d4a595b6d4b7
a=rtcp-fb:* transport-cc
a=rtcp-fb:* ccm fir
a=rtcp-fb:* goog-remb
a=rtcp-fb:* nack
a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01
a=mid:1


2024/02/12 16:22:13.347573 127.0.0.1:49088 -> 127.0.0.1:8080
ACK sip:127.0.0.1:8080;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.128;branch=z9hG4bK6900091
To: <sip:300@sip.myserv.com>;tag=c2c63d2d-76bf-4fc9-8394-d2807163001e
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
CSeq: 2 ACK
Call-ID: c0m4pdnp0q083l8hbsgu
Max-Forwards: 70
Supported: outbound
User-Agent: Browser Phone 0.3.24 (SIPJS - 0.20.0) Mozilla/5.0 (Linux; Android 10; K) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/121.0.0.0 Mobile Safari/537.36
Content-Length: 0



2024/02/12 16:22:20.118145 127.0.0.1:49088 -> 127.0.0.1:8080
BYE sip:127.0.0.1:8080;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.128;branch=z9hG4bK5883864
To: <sip:300@sip.myserv.com>;tag=c2c63d2d-76bf-4fc9-8394-d2807163001e
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
CSeq: 3 BYE
Call-ID: c0m4pdnp0q083l8hbsgu
Max-Forwards: 70
Supported: outbound
User-Agent: Browser Phone 0.3.24 (SIPJS - 0.20.0) Mozilla/5.0 (Linux; Android 10; K) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/121.0.0.0 Mobile Safari/537.36
Content-Length: 0



2024/02/12 16:22:20.118424 127.0.0.1:8080 -> 127.0.0.1:49088
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.128;rport=49088;received=127.0.0.1;branch=z9hG4bK5883864
Call-ID: c0m4pdnp0q083l8hbsgu
From: "Phone" <sip:User1@sip.myserv.com>;tag=2p03qrfok2
To: <sip:300@sip.myserv.com>;tag=c2c63d2d-76bf-4fc9-8394-d2807163001e
CSeq: 3 BYE
Server: Asterisk PBX 20.6.0
Content-Length:  0



2024/02/12 16:22:12.362182 192.168.58.6:5060 -> 192.168.102.22:5060
INVITE sip:User3@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPjec0af9d8-6f45-4fdc-a832-6696bd198b8b
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>
Contact: <sip:asterisk@192.168.58.6:5060>
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1092 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/sdp
Content-Length:   403

v=0
o=- 693107900 693107900 IN IP4 192.168.58.6
s=Asterisk
c=IN IP4 192.168.58.6
t=0 0
m=audio 11670 RTP/AVP 8 0 101
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
m=video 10726 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 packetization-mode=1;level-asymmetry-allowed=1;profile-level-id=42E01F
a=sendrecv


2024/02/12 16:22:12.396146 192.168.102.22:5060 -> 192.168.58.6:5060
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.58.6:5060;rport=5060;branch=z9hG4bKPjec0af9d8-6f45-4fdc-a832-6696bd198b8b
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1092 INVITE
User-Agent: BasIpVoip v2.0 (aa-07fbv2m, 3.22.1)
Content-Length: 0



2024/02/12 16:22:12.420189 192.168.102.22:5060 -> 192.168.58.6:5060
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.58.6:5060;rport=5060;branch=z9hG4bKPjec0af9d8-6f45-4fdc-a832-6696bd198b8b
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: "P-1-0-2" <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1092 INVITE
Contact: <sip:User3@192.168.102.22:5060>
Content-Type: application/sdp
User-Agent: BasIpVoip v2.0 (aa-07fbv2m, 3.22.1)
Content-Length:   335

v=0
o=dnake 1367547275 1367547275 IN IP4 192.168.102.22
s=dnake
c=IN IP4 192.168.102.22
t=0 0
m=audio 6100 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42001F; packetization-mode=1
a=sendrecv


2024/02/12 16:22:13.293986 192.168.102.22:5060 -> 192.168.58.6:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.58.6:5060;rport=5060;branch=z9hG4bKPjec0af9d8-6f45-4fdc-a832-6696bd198b8b
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: "P-1-0-2" <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1092 INVITE
Contact: <sip:1009902@192.168.102.22:5060>
Content-Type: application/sdp
User-Agent: BasIpVoip v2.0 (aa-07fbv2m, 3.22.1)
X-BAS-CALL-SOS: false
X-BAS-ELEVATOR-DTMF: DC
X-BAS-LOCK-DTMF-1: #
X-BAS-LOCK-DTMF-2: 0
X-BAS-LOCK-DTMF-ALL: *
Content-Length:   335

v=0
o=dnake 1367547275 1367547275 IN IP4 192.168.102.22
s=dnake
c=IN IP4 192.168.102.22
t=0 0
m=audio 6100 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000/1
a=rtpmap:101 telephone-event/8000/1
a=fmtp:101 0-11
a=sendrecv
m=video 6200 RTP/AVP 99
a=rtpmap:99 H264/90000
a=fmtp:99 profile-level-id=42001F; packetization-mode=1
a=sendrecv


2024/02/12 16:22:13.294583 192.168.58.6:5060 -> 192.168.102.22:5060
ACK sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj554b962f-0139-456c-a9b4-ac8aa9761845
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1092 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Length:  0



2024/02/12 16:22:13.970694 192.168.58.6:5060 -> 192.168.102.22:5060
INFO sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj07519626-b3c2-42f5-ac1f-b083cdd5f958
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1093 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>


2024/02/12 16:22:14.138631 192.168.102.22:5060 -> 192.168.58.6:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.58.6:5060;rport=5060;branch=z9hG4bKPj07519626-b3c2-42f5-ac1f-b083cdd5f958
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1093 INFO
Contact: <sip:User3@192.168.102.22:5060>
User-Agent: BasIpVoip v2.0 (aa-07fbv2m, 3.22.1)
Content-Length: 0



2024/02/12 16:22:14.297573 192.168.58.6:5060 -> 192.168.102.22:5060
INFO sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj4a52cc18-72e3-4229-a618-071282e7e721
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1094 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>


2024/02/12 16:22:14.505984 192.168.58.6:5060 -> 192.168.102.22:5060
INFO sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj66cbb837-2542-473a-b90d-334e06b5eeb8
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1095 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>


2024/02/12 16:22:14.683710 192.168.102.22:5060 -> 192.168.58.6:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.58.6:5060;rport=5060;branch=z9hG4bKPj66cbb837-2542-473a-b90d-334e06b5eeb8
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1095 INFO
Contact: <sip:User3@192.168.102.22:5060>
User-Agent: BasIpVoip v2.0 (aa-07fbv2m, 3.22.1)
Content-Length: 0



2024/02/12 16:22:14.704001 192.168.58.6:5060 -> 192.168.102.22:5060
INFO sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj37b664ff-dcaa-424e-9cf8-4ee0071d55d9
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1096 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>


2024/02/12 16:22:14.797914 192.168.58.6:5060 -> 192.168.102.22:5060
INFO sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj4a52cc18-72e3-4229-a618-071282e7e721
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1094 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>


2024/02/12 16:22:14.854642 192.168.102.22:5060 -> 192.168.58.6:5060
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.58.6:5060;rport=5060;branch=z9hG4bKPj4a52cc18-72e3-4229-a618-071282e7e721
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1094 INFO
Contact: <sip:User3@192.168.102.22:5060>
User-Agent: BasIpVoip v2.0 (aa-07fbv2m, 3.22.1)
Content-Length: 0



2024/02/12 16:22:14.906965 192.168.58.6:5060 -> 192.168.102.22:5060
INFO sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj196ed9e6-08a4-4036-8149-0a686522413c
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1097 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>


2024/02/12 16:22:15.109234 192.168.58.6:5060 -> 192.168.102.22:5060
INFO sip:1009902@192.168.102.22:5060 SIP/2.0
Via: SIP/2.0/UDP 192.168.58.6:5060;rport;branch=z9hG4bKPj5208e243-8f9e-4bc7-8164-63056321381f
From: "One Hundred" <sip:100@my_server_ip>;tag=18eba483-93f8-419d-8de5-a071e79f23fc
To: <sip:User3@192.168.102.22>;tag=443352001
Call-ID: 12f3a2ff-31a5-4f22-ac94-89af6b00b700
CSeq: 1098 INFO
Max-Forwards: 70
User-Agent: Asterisk PBX 20.6.0
Content-Type: application/media_control+xml
Content-Length:   178

<?xml version="1.0" encoding="utf-8" ?>
 <media_control>
  <vc_primitive>
   <to_encoder>
    <picture_fast_update/>
   </to_encoder>
  </vc_primitive>
 </media_control>


192.168.102.22 - is ip of intercom
192.168.58.0 - is my vpn network, 192.168.58.6 - is my asterisk server ip
User1 - webRTC client, User3(300) - intercom panel

Your log is out of sequence, and doesn’t include the dialplan execution.

This is asterisk shows in terminal:

  == Spawn extension (from-extensions, 300, 1) exited non-zero on 'PJSIP/User1-00000002'
    -- Executing [300@from-extensions:1] Dial("PJSIP/User1-00000004", "PJSIP/User3,30") in new stack
    -- Called PJSIP/User3
    -- PJSIP/User3-00000005 is making progress passing it to PJSIP/User1-00000004
    -- PJSIP/User1-00000004 requested media update control 26, passing it to PJSIP/User3-00000005
    -- PJSIP/User3-00000005 answered PJSIP/User1-00000004
    -- Channel PJSIP/User3-00000005 joined 'simple_bridge' basic-bridge <296f118b-6f1f-480a-ac08-7177b4a1e16a>
    -- Channel PJSIP/User1-00000004 joined 'simple_bridge' basic-bridge <296f118b-6f1f-480a-ac08-7177b4a1e16a>
[Feb 12 18:25:53] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:53] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:53] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:53] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:54] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
[Feb 12 18:25:55] WARNING[5342][C-00000003]: res_srtp.c:508 ast_srtp_protect: SRTP protect: replay check failed (index too old)
    -- Channel PJSIP/User1-00000004 left 'simple_bridge' basic-bridge <296f118b-6f1f-480a-ac08-7177b4a1e16a>
  == Spawn extension (from-extensions, 300, 1) exited non-zero on 'PJSIP/User1-00000004'
    -- Channel PJSIP/User3-00000005 left 'simple_bridge' basic-bridge <296f118b-6f1f-480a-ac08-7177b4a1e16a>

Here is full log of one call
https://pastebin.com/3nb17F48

https://csrc.nist.gov/glossary/term/replay_attack

Or the SRTP is otherwise corrupt.

Does this sound like there is a problem with the intercom video panel?

I have experienced the same problem, or sometimes I get a second or two of video before this occurs.

The error was being raised in res_srtp. Yes, this is a part of SRTP. It may not be corrupt, but out of date.

In my case, I didn’t trace down my error, I’m yet to solve the issue.

I was just trying to set it up according to your guide, calls between webRTC clients go through without problems, but there is a problem with the intercom

I noticed that when the webRTC client tries to receive a video stream from the intercom panel, many “INFO” requests come with the same content