PJSIP Asterisk and endpoints behind NAT

Hi,
I have read many topics, without getting any results.

The configuration is:
softphone ↔ (Mobile SIM: Router/NAT) ↔ Internet ↔ Router/NAT ↔ Asterisk & (LAN sip phones/ wss phones)

On router I added port forwaring UDP 5060 to asterisk and 10000-20000 (UDP) to asterisk for RTP.

My pjsip.conf:

[global]
max_forwards=70
user_agent=PBX VALERI
default_realm=pbx.valeri.it
keep_alive_interval=300

; == Transports
[transport-udp-nat]
type=transport
protocol=udp
bind=0.0.0.0
local_net=192.168.1.0/24
local_net=127.0.0.1/32
external_media_address=ddnsname.duckdns.org
external_signaling_address=ddnsname.duckdns.org

[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

[tcp_transport]
type=transport
protocol=tcp
bind=0.0.0.0
[tls_transport]
type=transport
protocol=tls
bind=0.0.0.0
cert_file=/root/certs/cert.pem
priv_key_file=/root/certs/cert.key
cipher=ADH-AES256-SHA,ADH-AES128-SHA
method=tlsv1

[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

; == 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,h263

[webrtc_endpoint](!)
transport=wss_transport
allow=opus,ulaw,vp9,vp8,h264,h263
; webrtc=yes
use_avpf=yes
media_encryption=dtls
dtls_verify=fingerprint
dtls_setup=actpass
ice_support=yes
media_use_received_transport=yes
rtcp_mux=yes
;dtls_cert_file=/root/certs/cert.pem
;dtls_private_key=/root/certs/cert.key
;dtls_ca_file=/root/ca/certs/ca.cert.pem

[test_endpoint](!)
;moh_suggest=default
context=from-extensions
use_avpf=yes
;inband_progress=no
allow_subscribe=yes
subscribe_context=subscriptions
direct_media=no
dtmf_mode=rfc4733
rtp_symmetric=yes
force_rport=yes
rewrite_contact=yes
ice_support=yes
rtp_keepalive=1
;device_state_busy_at=1
;disallow=all

; == Users
[Cucina](basic_endpoint,webrtc_endpoint)
type=endpoint
callerid="Cucina" <13>
auth=Cucina
aors=Cucina

[Cucina](single_aor)
type=aor

[Cucina](userpass_auth)
type=auth
username=Cucina
password=********

[TabletCucina](basic_endpoint,webrtc_endpoint)
type=endpoint
callerid="TabletCucina" <130>
auth=TabletCucina
aors=TabletCucina

[TabletCucina](single_aor)
type=aor

[TabletCucina](userpass_auth)
type=auth
username=TabletCucina
password=********

[COSTANTINO](test_endpoint, phone_endpoint)
type=endpoint
transport=transport-udp-nat
callerid="COSTANTINO" <901>
auth=COSTANTINO
aors=COSTANTINO

[COSTANTINO](single_aor)
type=aor

[COSTANTINO](userpass_auth)
type=auth
username=COSTANTINO
password=********

[DEBUG](basic_endpoint,webrtc_endpoint)
type=endpoint
callerid="DEBUG" <900>
auth=DEBUG
aors=DEBUG

[DEBUG](single_aor)
type=aor

[DEBUG](userpass_auth)
type=auth
username=DEBUG
password=********

My extensions.conf

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

[globals]
ATTENDED_TRANSFER_COMPLETE_SOUND=beep

[textmessages]
exten => 13,1,Gosub(send-text,s,1,(Cucina))
exten => 130,1,Gosub(send-text,s,1,(TabletCucina))
exten => 901,1,Gosub(send-text,s,1,(COSTANTINO))
exten => 900,1,Gosub(send-text,s,1,(DEBUG))

[subscriptions]
exten => 13,hint,PJSIP/Cucina
exten => 130,hint,PJSIP/TabletCucina
exten => 900,hint,PJSIP/DEBUG
exten => 901,hint,PJSIP/COSTANTINO

[from-extensions]
; Feature Codes:
exten => *65,1,Gosub(moh,s,1)
; Extensions
exten => 13,1,Gosub(dial-extension,s,1,(Cucina))
exten => 130,1,Gosub(dial-extension,s,1,(TabletCucina))
exten => 900,1,Gosub(dial-extension,s,1,(DEBUG))
;exten => 901,1,Gosub(dial-extension,s,1,(COSTANTINO))
exten => 901,1,Dial(PJSIP/COSTANTINO)
exten => 1000,1,Dial(PJSIP/TabletCucina&PJSIP/TabletCucina&PJSIP/DEBUG)
exten => 1200,1,Dial(PJSIP/DEBUG)

; Anything else, Hangup
exten => _[+*0-9].,1,NoOp(You called: ${EXTEN})
exten => _[+*0-9].,n,Hangup(1)
exten => e,1,Hangup()

[moh]
exten => s,1,NoOp(Music On Hold)
exten => s,n,Ringing()
exten => s,n,Wait(2)
exten => s,n,Answer()
exten => s,n,Wait(1)
exten => s,n,MusicOnHold()

[dial-extension]
exten => s,1,NoOp(Calling: ${ARG1})
exten => s,n,Set(JITTERBUFFER(adaptive)=default)
exten => s,n,Dial(PJSIP/${ARG1},30)
exten => s,n,Hangup()
exten => e,1,Hangup()

[send-text]
exten => s,1,NoOp(Sending Text To: ${ARG1})
exten => s,n,Set(PEER=${CUT(CUT(CUT(MESSAGE(from),@,1),<,2),:,2)})
exten => s,n,Set(FROM=${SHELL(asterisk -rx 'pjsip show endpoint ${PEER}' | grep 'callerid ' | cut -d':' -f2- | sed 's/^\ *//' | tr -d '\n')})
exten => s,n,Set(CALLERID_NUM=${CUT(CUT(FROM,>,1),<,2)})
exten => s,n,Set(FROM_SIP=${STRREPLACE(MESSAGE(from),<sip:${PEER}@,<sip:${CALLERID_NUM}@)})
exten => s,n,MessageSend(pjsip:${ARG1},${FROM_SIP})
exten => s,n,Hangup()

This is the asterisk cli:

Attempted to remove non-existent contact 'sip:COSTANTINO@46.44.193.151:47845;x-ast-orig-host=10.139.1.218:47845' from AOR 'COSTANTINO' by request
    -- Added contact 'sip:COSTANTINO@46.44.193.151:47845;x-ast-orig-host=10.139.1.218:47845' to AOR 'COSTANTINO' with expiration of 3600 seconds
  == Endpoint COSTANTINO is now Reachable
    -- Contact COSTANTINO/sip:COSTANTINO@46.44.193.151:47845;x-ast-orig-host=10.139.1.218:47845 is now Reachable.  RTT: 165.158 msec

remote phone successfully registration

== Setting global variable 'SIPDOMAIN' to 'pbx.valeri.it'
    -- Executing [901@from-extensions:1] Dial("PJSIP/DEBUG-0000005a", "PJSIP/COSTANTINO") in new stack
  == Setting global variable 'SIPDOMAIN' to 'pbx.valeri.it'
    -- Called PJSIP/COSTANTINO
    -- Executing [901@from-extensions:1] Dial("PJSIP/DEBUG-0000005c", "PJSIP/COSTANTINO") in new stack
    -- Called PJSIP/COSTANTINO
<< [ TYPE: Control (4) SUBCLASS: Unknown control '22' (22) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '22' (22) ] [PJSIP/COSTANTINO-0000005d]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [PJSIP/COSTANTINO-0000005b]
    -- PJSIP/COSTANTINO-0000005b is ringing
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [PJSIP/COSTANTINO-0000005b]
    -- PJSIP/COSTANTINO-0000005b is ringing
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005d]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005d]
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [PJSIP/COSTANTINO-0000005d]
    -- PJSIP/COSTANTINO-0000005d is ringing
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005d]
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [PJSIP/COSTANTINO-0000005d]
    -- PJSIP/COSTANTINO-0000005d is ringing
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [PJSIP/COSTANTINO-0000005b]
    -- PJSIP/COSTANTINO-0000005b answered PJSIP/DEBUG-0000005a
    -- Channel PJSIP/COSTANTINO-0000005b joined 'simple_bridge' basic-bridge <2be157c9-b0be-455d-bdba-149bbd942b02>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/COSTANTINO-0000005b]
    -- Channel PJSIP/DEBUG-0000005a joined 'simple_bridge' basic-bridge <2be157c9-b0be-455d-bdba-149bbd942b02>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/DEBUG-0000005a]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/DEBUG-0000005a]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/COSTANTINO-0000005b]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/COSTANTINO-0000005b]
<< [ HANGUP (NULL) ] [PJSIP/DEBUG-0000005a]
    -- Channel PJSIP/DEBUG-0000005a left 'simple_bridge' basic-bridge <2be157c9-b0be-455d-bdba-149bbd942b02>
    -- Channel PJSIP/COSTANTINO-0000005b left 'simple_bridge' basic-bridge <2be157c9-b0be-455d-bdba-149bbd942b02>
  == Spawn extension (from-extensions, 901, 1) exited non-zero on 'PJSIP/DEBUG-0000005a'

LAN phone calls correctly remote softphone and it rings but on answer the call breaks out.
Can you help me?

You would need to show an actual SIP trace using “pjsip set logger on” at what the signaling is doing. Configuration should also be put on here using the “preformatted text” option of the editor so it is unmodified/unstyled.

I formatted my topic. This is the logger during the outbound call (pjsip set logger on and rtp set debug on):

<--- Received SIP request (4058 bytes) from WSS:192.168.1.125:47752 --->
INVITE sip:901@pbx.valeri.it SIP/2.0
Via: SIP/2.0/WSS crqm9abjklrc.invalid;branch=z9hG4bK1691087
Max-Forwards: 69
To: <sip:901@pbx.valeri.it>
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
Call-ID: nhkgq7p0639b9b81hl65
CSeq: 1337 INVITE
Contact: <sip:mpkmesmc@crqm9abjklrc.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSip-DEBUG
Content-Length: 3540

v=0
o=- 3213490020041157574 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 60116a08-62bd-48d3-8d01-34e90703b06e
m=audio 61903 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 192.168.1.125
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2260986139 1 udp 2122260223 192.168.1.125 61903 typ host generation 0 network-id 7 network-cost 50
a=candidate:1100181702 1 udp 2122194687 169.254.33.89 61904 typ host generation 0 network-id 1
a=candidate:2578558971 1 udp 2122129151 192.168.0.10 61905 typ host generation 0 network-id 2
a=candidate:2263023274 1 udp 2122063615 172.19.144.1 61906 typ host generation 0 network-id 3
a=candidate:1032414325 1 udp 2121998079 172.30.48.1 61907 typ host generation 0 network-id 4
a=candidate:1508684856 1 udp 2121932543 172.25.48.1 61908 typ host generation 0 network-id 5
a=candidate:1470687749 1 udp 2121867007 172.17.160.1 61909 typ host generation 0 network-id 6
a=candidate:3580006164 1 udp 2121801471 172.21.16.1 61910 typ host generation 0 network-id 8
a=candidate:1582909429 1 udp 2121735935 169.254.142.95 61911 typ host generation 0 network-id 9
a=candidate:2619542049 1 udp 2121670399 169.254.116.155 61912 typ host generation 0 network-id 10
a=candidate:961251751 1 udp 2121604863 172.22.32.1 61913 typ host generation 0 network-id 11
a=candidate:4161538947 1 tcp 1518280447 192.168.1.125 9 typ host tcptype active generation 0 network-id 7 network-cost 50
a=candidate:1063031390 1 tcp 1518214911 169.254.33.89 9 typ host tcptype active generation 0 network-id 1
a=candidate:3883814243 1 tcp 1518149375 192.168.0.10 9 typ host tcptype active generation 0 network-id 2
a=candidate:4163705906 1 tcp 1518083839 172.19.144.1 9 typ host tcptype active generation 0 network-id 3
a=candidate:1128695533 1 tcp 1518018303 172.30.48.1 9 typ host tcptype active generation 0 network-id 4
a=candidate:656627360 1 tcp 1517952767 172.25.48.1 9 typ host tcptype active generation 0 network-id 5
a=candidate:694618269 1 tcp 1517887231 172.17.160.1 9 typ host tcptype active generation 0 network-id 6
a=candidate:2880269708 1 tcp 1517821695 172.21.16.1 9 typ host tcptype active generation 0 network-id 8
a=candidate:546743661 1 tcp 1517756159 169.254.142.95 9 typ host tcptype active generation 0 network-id 9
a=candidate:3807176889 1 tcp 1517690623 169.254.116.155 9 typ host tcptype active generation 0 network-id 10
a=candidate:1199860543 1 tcp 1517625087 172.22.32.1 9 typ host tcptype active generation 0 network-id 11
a=ice-ufrag:x8Xe
a=ice-pwd:ckAkBYUmJ6cwCMbY/Xzf3kwn
a=ice-options:trickle
a=fingerprint:sha-256 10:43:78:3C:BA:0B:AE:CA:DB:22:8D:F3:5F:B4:26:82:CE:CD:8A:C4:75:22:67:22:11:E1:77:9D:F0:4A:D3:00
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:60116a08-62bd-48d3-8d01-34e90703b06e b84ddc40-6685-4357-90e6-6e8ab0ff2281
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:3850452358 cname:U2PIXioVyHOg3ss3
a=ssrc:3850452358 msid:60116a08-62bd-48d3-8d01-34e90703b06e b84ddc40-6685-4357-90e6-6e8ab0ff2281

<--- Transmitting SIP response (461 bytes) to WSS:192.168.1.125:47752 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS crqm9abjklrc.invalid;rport=47752;received=192.168.1.125;branch=z9hG4bK1691087
Call-ID: nhkgq7p0639b9b81hl65
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
To: <sip:901@pbx.valeri.it>;tag=z9hG4bK1691087
CSeq: 1337 INVITE
WWW-Authenticate: Digest realm="pbx.valeri.it",nonce="1695287401/a7eed640fb9769490e54bad4eba9dc99",opaque="65c3fee34afa69ec",algorithm=md5,qop="auth"
Server: PBX VALERI
Content-Length:  0


<--- Received SIP request (395 bytes) from WSS:192.168.1.125:47752 --->
ACK sip:901@pbx.valeri.it SIP/2.0
Via: SIP/2.0/WSS crqm9abjklrc.invalid;branch=z9hG4bK1691087
Max-Forwards: 69
To: <sip:901@pbx.valeri.it>;tag=z9hG4bK1691087
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
Call-ID: nhkgq7p0639b9b81hl65
CSeq: 1337 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSip-DEBUG
Content-Length: 0


<--- Received SIP request (4335 bytes) from WSS:192.168.1.125:47752 --->
INVITE sip:901@pbx.valeri.it SIP/2.0
Via: SIP/2.0/WSS crqm9abjklrc.invalid;branch=z9hG4bK948077
Max-Forwards: 69
To: <sip:901@pbx.valeri.it>
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
Call-ID: nhkgq7p0639b9b81hl65
CSeq: 1338 INVITE
Authorization: Digest algorithm=MD5, username="DEBUG", realm="pbx.valeri.it", nonce="1695287401/a7eed640fb9769490e54bad4eba9dc99", uri="sip:901@pbx.valeri.it", response="9750a04b0df12202aef4d0bf08763813", opaque="65c3fee34afa69ec", qop=auth, cnonce="g0adt4e5ojgc", nc=00000001
Contact: <sip:mpkmesmc@crqm9abjklrc.invalid;transport=ws;ob>
Content-Type: application/sdp
Session-Expires: 90
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: timer,ice,replaces,outbound
User-Agent: JsSip-DEBUG
Content-Length: 3540

v=0
o=- 3213490020041157574 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 60116a08-62bd-48d3-8d01-34e90703b06e
m=audio 61903 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 192.168.1.125
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2260986139 1 udp 2122260223 192.168.1.125 61903 typ host generation 0 network-id 7 network-cost 50
a=candidate:1100181702 1 udp 2122194687 169.254.33.89 61904 typ host generation 0 network-id 1
a=candidate:2578558971 1 udp 2122129151 192.168.0.10 61905 typ host generation 0 network-id 2
a=candidate:2263023274 1 udp 2122063615 172.19.144.1 61906 typ host generation 0 network-id 3
a=candidate:1032414325 1 udp 2121998079 172.30.48.1 61907 typ host generation 0 network-id 4
a=candidate:1508684856 1 udp 2121932543 172.25.48.1 61908 typ host generation 0 network-id 5
a=candidate:1470687749 1 udp 2121867007 172.17.160.1 61909 typ host generation 0 network-id 6
a=candidate:3580006164 1 udp 2121801471 172.21.16.1 61910 typ host generation 0 network-id 8
a=candidate:1582909429 1 udp 2121735935 169.254.142.95 61911 typ host generation 0 network-id 9
a=candidate:2619542049 1 udp 2121670399 169.254.116.155 61912 typ host generation 0 network-id 10
a=candidate:961251751 1 udp 2121604863 172.22.32.1 61913 typ host generation 0 network-id 11
a=candidate:4161538947 1 tcp 1518280447 192.168.1.125 9 typ host tcptype active generation 0 network-id 7 network-cost 50
a=candidate:1063031390 1 tcp 1518214911 169.254.33.89 9 typ host tcptype active generation 0 network-id 1
a=candidate:3883814243 1 tcp 1518149375 192.168.0.10 9 typ host tcptype active generation 0 network-id 2
a=candidate:4163705906 1 tcp 1518083839 172.19.144.1 9 typ host tcptype active generation 0 network-id 3
a=candidate:1128695533 1 tcp 1518018303 172.30.48.1 9 typ host tcptype active generation 0 network-id 4
a=candidate:656627360 1 tcp 1517952767 172.25.48.1 9 typ host tcptype active generation 0 network-id 5
a=candidate:694618269 1 tcp 1517887231 172.17.160.1 9 typ host tcptype active generation 0 network-id 6
a=candidate:2880269708 1 tcp 1517821695 172.21.16.1 9 typ host tcptype active generation 0 network-id 8
a=candidate:546743661 1 tcp 1517756159 169.254.142.95 9 typ host tcptype active generation 0 network-id 9
a=candidate:3807176889 1 tcp 1517690623 169.254.116.155 9 typ host tcptype active generation 0 network-id 10
a=candidate:1199860543 1 tcp 1517625087 172.22.32.1 9 typ host tcptype active generation 0 network-id 11
a=ice-ufrag:x8Xe
a=ice-pwd:ckAkBYUmJ6cwCMbY/Xzf3kwn
a=ice-options:trickle
a=fingerprint:sha-256 10:43:78:3C:BA:0B:AE:CA:DB:22:8D:F3:5F:B4:26:82:CE:CD:8A:C4:75:22:67:22:11:E1:77:9D:F0:4A:D3:00
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:60116a08-62bd-48d3-8d01-34e90703b06e b84ddc40-6685-4357-90e6-6e8ab0ff2281
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:3850452358 cname:U2PIXioVyHOg3ss3
a=ssrc:3850452358 msid:60116a08-62bd-48d3-8d01-34e90703b06e b84ddc40-6685-4357-90e6-6e8ab0ff2281

  == Setting global variable 'SIPDOMAIN' to 'pbx.valeri.it'
<--- Transmitting SIP response (284 bytes) to WSS:192.168.1.125:47752 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS crqm9abjklrc.invalid;rport=47752;received=192.168.1.125;branch=z9hG4bK948077
Call-ID: nhkgq7p0639b9b81hl65
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
To: <sip:901@pbx.valeri.it>
CSeq: 1338 INVITE
Server: PBX VALERI
Content-Length:  0


    -- Executing [901@from-extensions:1] Dial("PJSIP/DEBUG-0000006c", "PJSIP/COSTANTINO") in new stack
    -- Called PJSIP/COSTANTINO
<< [ TYPE: Control (4) SUBCLASS: Unknown control '22' (22) ] [PJSIP/COSTANTINO-0000006d]
<--- Transmitting SIP request (1763 bytes) to UDP:46.44.193.151:24200 --->
INVITE sip:COSTANTINO@46.44.193.151:24200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;rport;branch=z9hG4bKPj35a5d7e0-b6fa-47a9-935a-363db2a21d63
From: "DEBUG" <sip:900@192.168.1.50>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>
Contact: <sip:asterisk@192.168.1.50:5060>
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 764 INVITE
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: PBX VALERI
Content-Type: application/sdp
Content-Length:  1092

v=0
o=- 1416687995 1416687995 IN IP4 192.168.1.50
s=Asterisk
c=IN IP4 192.168.1.50
t=0 0
m=audio 14592 UDP/TLS/RTP/SAVPF 107 0 101
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
a=ice-ufrag:5758cec605d93ea71ed10fc517c571d5
a=ice-pwd:3b42c85b57d5937a1c1280211483b3c8
a=candidate:Hc0a80132 1 UDP 2130706431 192.168.1.50 14592 typ host
a=candidate:S5d979b61 1 UDP 1694498815 93.151.155.97 14592 typ srflx raddr 192.168.1.50 rport 14592
a=rtpmap:107 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux
m=video 14890 UDP/TLS/RTP/SAVPF 108 100 99 34
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
a=ice-ufrag:79f5d20e3e91d1024b155451359b1df4
a=ice-pwd:024b8c0617b64c370bdb076c3a634854
a=candidate:Hc0a80132 1 UDP 2130706431 192.168.1.50 14890 typ host
a=candidate:S5d979b61 1 UDP 1694498815 93.151.155.97 14890 typ srflx raddr 192.168.1.50 rport 14890
a=rtpmap:108 VP9/90000
a=rtpmap:100 VP8/90000
a=rtpmap:99 H264/90000
a=rtpmap:34 H263/90000
a=sendrecv
a=rtcp-mux

<--- Received SIP response (494 bytes) from UDP:46.44.193.151:24200 --->
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 192.168.1.50:5060;rport=5060;branch=z9hG4bKPj35a5d7e0-b6fa-47a9-935a-363db2a21d63
From: "DEBUG" <sip:900@192.168.1.50:5060>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 764 INVITE
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000006d]
<--- Received SIP response (588 bytes) from UDP:46.44.193.151:24200 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 192.168.1.50:5060;rport=5060;branch=z9hG4bKPj35a5d7e0-b6fa-47a9-935a-363db2a21d63
From: "DEBUG" <sip:900@192.168.1.50:5060>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 764 INVITE
Contact: <sip:COSTANTINO@10.139.1.218:24200>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow-Events: talk, hold
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000006d]
<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [PJSIP/COSTANTINO-0000006d]
    -- PJSIP/COSTANTINO-0000006d is ringing
<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000006d]
<--- Transmitting SIP response (484 bytes) to WSS:192.168.1.125:47752 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS crqm9abjklrc.invalid;rport=47752;received=192.168.1.125;branch=z9hG4bK948077
Call-ID: nhkgq7p0639b9b81hl65
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
To: <sip:901@pbx.valeri.it>;tag=c85d6178-dbee-451f-8a88-3dbc0aade033
CSeq: 1338 INVITE
Server: PBX VALERI
Contact: <sip:192.168.1.50:4443;transport=ws>
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Content-Length:  0


<< [ TYPE: Control (4) SUBCLASS: Ringing (3) ] [PJSIP/COSTANTINO-0000006d]
    -- PJSIP/COSTANTINO-0000006d is ringing
<--- Received SIP response (1378 bytes) from UDP:46.44.193.151:24200 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;rport=5060;branch=z9hG4bKPj35a5d7e0-b6fa-47a9-935a-363db2a21d63
From: "DEBUG" <sip:900@192.168.1.50:5060>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 764 INVITE
Contact: <sip:COSTANTINO@10.139.1.218:24200>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   733

v=0
o=COSTANTINO 8001 8000 IN IP4 10.139.1.218
s=SIP Call
c=IN IP4 10.139.1.218
t=0 0
m=audio 22380 RTP/AVP 0 101
a=sendrecv
a=rtcp:22381 IN IP4 10.139.1.218
a=rtpmap:0 PCMU/8000
a=ptime:10
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 UDP/TLS/RTP/SAVPF 108 100 99 34
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
a=ice-ufrag:79f5d20e3e91d1024b155451359b1df4
a=ice-pwd:024b8c0617b64c370bdb076c3a634854
a=candidate:Hc0a80132 1 UDP 2130706431 192.168.1.50 14890 typ host
a=candidate:S5d979b61 1 UDP 1694498815 93.151.155.97 14890 typ srflx raddr 192.168.1.50 rport 14890
a=rtpmap:108 VP9/90000
a=rtpmap:100 VP8/90000
a=rtpmap:99 H264/90000
a=rtpmap:34 H263/90000
a=sendrecv
a=rtcp-mux

<< [ TYPE: Control (4) SUBCLASS: Unknown control '33' (33) ] [PJSIP/COSTANTINO-0000006d]
<< [ TYPE: Control (4) SUBCLASS: Answer (4) ] [PJSIP/COSTANTINO-0000006d]
    -- PJSIP/COSTANTINO-0000006d answered PJSIP/DEBUG-0000006c
<--- Transmitting SIP response (1209 bytes) to WSS:192.168.1.125:47752 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS crqm9abjklrc.invalid;rport=47752;received=192.168.1.125;branch=z9hG4bK948077
Call-ID: nhkgq7p0639b9b81hl65
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
To: <sip:901@pbx.valeri.it>;tag=c85d6178-dbee-451f-8a88-3dbc0aade033
CSeq: 1338 INVITE
Server: PBX VALERI
Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REGISTER, REFER, MESSAGE
Contact: <sip:192.168.1.50:4443;transport=ws>
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 90;refresher=uac
Require: timer
Content-Type: application/sdp
Content-Length:   597

v=0
o=- 3184585670 4 IN IP4 192.168.1.50
s=Asterisk
c=IN IP4 192.168.1.50
t=0 0
m=audio 14132 UDP/TLS/RTP/SAVPF 111 0 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256
a=ice-ufrag:178df83611067cc15cc059f970a8ca8e
a=ice-pwd:5c1fee7a335f48572cb7cbca2469c819
a=candidate:Hc0a80132 1 UDP 2130706431 192.168.1.50 14132 typ host
a=candidate:S5d979b61 1 UDP 1694498815 93.151.155.97 14132 typ srflx raddr 192.168.1.50 rport 14132
a=rtpmap:111 opus/48000/2
a=rtpmap:0 PCMU/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
a=rtcp-mux

    -- Channel PJSIP/COSTANTINO-0000006d joined 'simple_bridge' basic-bridge <62b658c7-4bf7-4e9b-aa68-f7c385dc61f6>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/COSTANTINO-0000006d]
    -- Channel PJSIP/DEBUG-0000006c joined 'simple_bridge' basic-bridge <62b658c7-4bf7-4e9b-aa68-f7c385dc61f6>
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/DEBUG-0000006c]
<< [ TYPE: Null Frame (5) SUBCLASS: N/A (0) ] [PJSIP/DEBUG-0000006c]
<--- Received SIP request (430 bytes) from WSS:192.168.1.125:47752 --->
ACK sip:192.168.1.50:4443;transport=ws SIP/2.0
Via: SIP/2.0/WSS crqm9abjklrc.invalid;branch=z9hG4bK3877629
Max-Forwards: 69
To: <sip:901@pbx.valeri.it>;tag=c85d6178-dbee-451f-8a88-3dbc0aade033
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
Call-ID: nhkgq7p0639b9b81hl65
CSeq: 1338 ACK
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSip-DEBUG
Content-Length: 0


<--- Received SIP request (481 bytes) from WSS:192.168.1.125:47752 --->
BYE sip:192.168.1.50:4443;transport=ws SIP/2.0
Via: SIP/2.0/WSS crqm9abjklrc.invalid;branch=z9hG4bK259055
Max-Forwards: 69
To: <sip:901@pbx.valeri.it>;tag=c85d6178-dbee-451f-8a88-3dbc0aade033
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
Call-ID: nhkgq7p0639b9b81hl65
CSeq: 1339 BYE
Reason: SIP ;cause=488; text="Not Acceptable Here"
Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
Supported: outbound
User-Agent: JsSip-DEBUG
Content-Length: 0


<--- Transmitting SIP response (318 bytes) to WSS:192.168.1.125:47752 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS crqm9abjklrc.invalid;rport=47752;received=192.168.1.125;branch=z9hG4bK259055
Call-ID: nhkgq7p0639b9b81hl65
From: <sip:DEBUG@pbx.valeri.it>;tag=de9ruhh3s7
To: <sip:901@pbx.valeri.it>;tag=c85d6178-dbee-451f-8a88-3dbc0aade033
CSeq: 1339 BYE
Server: PBX VALERI
Content-Length:  0


<< [ HANGUP (NULL) ] [PJSIP/DEBUG-0000006c]
    -- Channel PJSIP/DEBUG-0000006c left 'simple_bridge' basic-bridge <62b658c7-4bf7-4e9b-aa68-f7c385dc61f6>
    -- Channel PJSIP/COSTANTINO-0000006d left 'simple_bridge' basic-bridge <62b658c7-4bf7-4e9b-aa68-f7c385dc61f6>
  == Spawn extension (from-extensions, 901, 1) exited non-zero on 'PJSIP/DEBUG-0000006c'
<--- Transmitting SIP request (398 bytes) to UDP:46.44.193.151:24200 --->
ACK sip:COSTANTINO@46.44.193.151:24200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;rport;branch=z9hG4bKPj1ac8e719-7361-4171-97f4-c45aa1e1fa28
From: "DEBUG" <sip:900@192.168.1.50>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 764 ACK
Max-Forwards: 70
User-Agent: PBX VALERI
Content-Length:  0


<--- Transmitting SIP request (422 bytes) to UDP:46.44.193.151:24200 --->
BYE sip:COSTANTINO@46.44.193.151:24200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;rport;branch=z9hG4bKPjf854aa3e-7bb5-4b83-9433-ce0f13747f44
From: "DEBUG" <sip:900@192.168.1.50>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 765 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: PBX VALERI
Content-Length:  0


<--- Received SIP response (1378 bytes) from UDP:46.44.193.151:24200 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;rport=5060;branch=z9hG4bKPj35a5d7e0-b6fa-47a9-935a-363db2a21d63
From: "DEBUG" <sip:900@192.168.1.50:5060>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 764 INVITE
Contact: <sip:COSTANTINO@10.139.1.218:24200>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Session-Expires: 1800;refresher=uac
Require: timer
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Type: application/sdp
Content-Length:   733

v=0
o=COSTANTINO 8001 8000 IN IP4 10.139.1.218
s=SIP Call
c=IN IP4 10.139.1.218
t=0 0
m=audio 22380 RTP/AVP 0 101
a=sendrecv
a=rtcp:22381 IN IP4 10.139.1.218
a=rtpmap:0 PCMU/8000
a=ptime:10
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
m=video 0 UDP/TLS/RTP/SAVPF 108 100 99 34
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256
a=ice-ufrag:79f5d20e3e91d1024b155451359b1df4
a=ice-pwd:024b8c0617b64c370bdb076c3a634854
a=candidate:Hc0a80132 1 UDP 2130706431 192.168.1.50 14890 typ host
a=candidate:S5d979b61 1 UDP 1694498815 93.151.155.97 14890 typ srflx raddr 192.168.1.50 rport 14890
a=rtpmap:108 VP9/90000
a=rtpmap:100 VP8/90000
a=rtpmap:99 H264/90000
a=rtpmap:34 H263/90000
a=sendrecv
a=rtcp-mux

<--- Transmitting SIP request (398 bytes) to UDP:46.44.193.151:24200 --->
ACK sip:COSTANTINO@46.44.193.151:24200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;rport;branch=z9hG4bKPj1ac8e719-7361-4171-97f4-c45aa1e1fa28
From: "DEBUG" <sip:900@192.168.1.50>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 764 ACK
Max-Forwards: 70
User-Agent: PBX VALERI
Content-Length:  0


<--- Transmitting SIP request (452 bytes) to UDP:192.168.1.12:35537 --->
OPTIONS sip:TabletIngresso@192.168.1.12:35537 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;rport;branch=z9hG4bKPj8e7fc32d-0855-40cc-b2c3-7154dca93992
From: <sip:TabletIngresso@192.168.1.50>;tag=136fccd1-b539-4beb-b748-314ed4d87e4c
To: <sip:TabletIngresso@192.168.1.12>
Contact: <sip:TabletIngresso@192.168.1.50:5060>
Call-ID: 0ef12bdc-b7b1-4bd4-bd30-ee15c1293159
CSeq: 48447 OPTIONS
Max-Forwards: 70
User-Agent: PBX VALERI
Content-Length:  0


<--- Received SIP response (509 bytes) from UDP:192.168.1.12:35537 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;rport=5060;branch=z9hG4bKPj8e7fc32d-0855-40cc-b2c3-7154dca93992
From: <sip:TabletIngresso@192.168.1.50>;tag=136fccd1-b539-4beb-b748-314ed4d87e4c
To: <sip:TabletIngresso@192.168.1.12>;tag=1061781312
Call-ID: 0ef12bdc-b7b1-4bd4-bd30-ee15c1293159
CSeq: 48447 OPTIONS
Supported: replaces, path, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


Sent Comfort Noise RTP packet to 10.139.1.218:22380 (type 02, seq 025033, ts 000000, len 000001)
<--- Transmitting SIP request (422 bytes) to UDP:46.44.193.151:24200 --->
BYE sip:COSTANTINO@46.44.193.151:24200 SIP/2.0
Via: SIP/2.0/UDP 192.168.1.50:5060;rport;branch=z9hG4bKPjf854aa3e-7bb5-4b83-9433-ce0f13747f44
From: "DEBUG" <sip:900@192.168.1.50>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 765 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: PBX VALERI
Content-Length:  0


<--- Received SIP response (554 bytes) from UDP:46.44.193.151:24200 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;rport=5060;branch=z9hG4bKPjf854aa3e-7bb5-4b83-9433-ce0f13747f44
From: "DEBUG" <sip:900@192.168.1.50:5060>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 765 BYE
Contact: <sip:COSTANTINO@10.139.1.218:24200>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0


<--- Received SIP response (554 bytes) from UDP:46.44.193.151:24200 --->
SIP/2.0 200 OK
Via: SIP/2.0/UDP 192.168.1.50:5060;rport=5060;branch=z9hG4bKPjf854aa3e-7bb5-4b83-9433-ce0f13747f44
From: "DEBUG" <sip:900@192.168.1.50:5060>;tag=ac899419-ab61-47cf-a97c-22bfb0d1d0a5
To: <sip:COSTANTINO@46.44.193.151>;tag=318245857
Call-ID: 6425da76-5dbd-4672-9834-e26a2ca2e004
CSeq: 765 BYE
Contact: <sip:COSTANTINO@10.139.1.218:24200>
Supported: replaces, path, timer, eventlist
User-Agent: Grandstream Wave 1.0.3.34
Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, INFO, REFER, UPDATE, MESSAGE
Content-Length: 0

JsSIP has rejected the SDP from Asterisk for some reason. It would appear to be due to DTLS. According to the configuration you haven’t configured it with a certificate which is required, or enabled the auto certificate generation. In general the “webrtc” option should just be used which configures an endpoint for WebRTC.

All right.
In fact, calling from a SIP LAN phone to a remote phone makes the call communication is established, but no audio and/or video received.

You need to pick a scenario to diagnose with and make work first, without including yet another scenario. If the change caused the WebRTC case to work - then state so and move on to providing new information for the new scenario.

Ok, I will create a new topic for other scenario. Thank you for your support

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