Directmedia not working when devices and asterisk server is in local network

Asterisk Asterisk SIP
1

Hello,
I have to make P2P call between 2 client application.
Both the applications are running in my window PC on 2 different chrome browser instances(one instance is in incognito mode).
I am accessing Asterisk through VPN.
My PC IP address in 192.168.6.100 and Asterisk IP is 192.168.4.189

I tried using directmedia in sip.conf, but didn’t get any success in establishing P2P call.
I have also gone through almost all the asterisk topics/threads related to directmedia, but that also didn’t help.

Below are my sip.conf settings
[general]
auth_message_requests=no
udpbindaddr=0.0.0.0:5060
ignoresdpversion=yes
realm=raspberrypi.local
allowguest=yes
videosupport=no

jbenable=no
dtmfmode=info
transport=udp
permit=192.168.0.0/255.255.0.0
directmediapermit=192.168.6.0/255.255.0.0
directmediapermit=192.168.4.0/255.255.0.0
rtcp_mux=yes

; == Templates

basic
type=peer
context=from-extensions
host=dynamic
directmedia=yes
nat=never
dtmfmode=info
disallow=all
videosupport=no

webrtc
transport=wss,udp
videosupport=no
allow=opus
avpf=yes
icesupport=yes
rtcp_mux=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/etc/asterisk/keys/asterisk.pem
dtlscafile=/etc/asterisk/keys/asterisk.crt
dtlssetup=actpass

; == Users
User1
callerid=“User 1” <100>
;secret=1234

User2
callerid=“User 2” <200>
;secret=1234

Extension.conf settings
[general]

[globals]

[from-extensions]
; Extensions
exten => 100,1,Dial(SIP/User1,30)
exten => 200,1,Dial(SIP/User2,30)

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

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

Sip Logs

<— SIP read from WS:192.168.6.100:60326 —>
INVITE sip:100@192.168.4.189 SIP/2.0
Via: SIP/2.0/WSS 192.0.2.123;branch=z9hG4bK4796544
Max-Forwards: 70
To: sip:100@192.168.4.189
From: “MyApp” sip:User2@192.168.4.189;tag=vlk9qceijm
Call-ID: 4dg3sdggf4fnq0f0v8ua
CSeq: 1751 INVITE
Contact: sip:06r3i1vi@192.0.2.123;transport=wss;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: MyApp App
Content-Type: application/sdp
Content-Length: 2192

v=0
o=- 8287664629104701326 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS JLFvoW8tMQZuZ9MAHhep6eh7odYDh1eC3LCF
m=audio 62748 UDP/TLS/RTP/SAVPF 111
c=IN IP4 192.168.6.100
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3110093121 1 udp 2122260223 192.168.6.100 62748 typ host generation 0 network-id 5 network-cost 50
a=candidate:633891456 1 udp 2122189567 2001::348b:fb58:1401:1a78:62d2:af60 62749 typ host generation 0 network-id 4 network-cost 50
a=candidate:4027572201 1 udp 2122131711 2409:4071:208d:e6fd:952a:2959:97f3:b9d9 62750 typ host generation 0 network-id 2 network-cost 10
a=candidate:951146822 1 udp 2122066175 2409:4071:208d:e6fd:a83e:c7e2:fb68:a2e9 62751 typ host generation 0 network-id 3 network-cost 10
a=candidate:419066100 1 udp 2121998079 192.168.225.28 62752 typ host generation 0 network-id 1 network-cost 10
a=candidate:4158897585 1 tcp 1518280447 192.168.6.100 9 typ host tcptype active generation 0 network-id 5 network-cost 50
a=candidate:1800115824 1 tcp 1518209791 2001::348b:fb58:1401:1a78:62d2:af60 9 typ host tcptype active generation 0 network-id 4 network-cost 50
a=candidate:3196855065 1 tcp 1518151935 2409:4071:208d:e6fd:952a:2959:97f3:b9d9 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:1983030710 1 tcp 1518086399 2409:4071:208d:e6fd:a83e:c7e2:fb68:a2e9 9 typ host tcptype active generation 0 network-id 3 network-cost 10
a=candidate:1450806276 1 tcp 1518018303 192.168.225.28 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:dzeI
a=ice-pwd:Aa3rzu29j19yqoRD0NcGeP6C
a=fingerprint:sha-256 27:78:58:BB:88:0C:1B:47:DF:DA:06:67:09:E1:78:2C:50:28:0B:FC:44:2D:39:D1:4A:A0:91:8D:08:5D:F3:B7
a=setup:actpass
a=mid:0
a=sendrecv
a=msid:JLFvoW8tMQZuZ9MAHhep6eh7odYDh1eC3LCF 267821bc-6223-458f-a51f-93e89f1ccea7
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 useinbandfec=1
a=maxptime:20
a=ssrc:998397670 cname:j55aLdaqzCO9KIks
a=ssrc:998397670 msid:JLFvoW8tMQZuZ9MAHhep6eh7odYDh1eC3LCF 267821bc-6223-458f-a51f-93e89f1ccea7
a=ssrc:998397670 mslabel:JLFvoW8tMQZuZ9MAHhep6eh7odYDh1eC3LCF
a=ssrc:998397670 label:267821bc-6223-458f-a51f-93e89f1ccea7
<------------->
— (13 headers 34 lines) —
Using INVITE request as basis request - 4dg3sdggf4fnq0f0v8ua
Found peer ‘User2’ for ‘User2’ from 192.168.6.100:60326
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP RTP CoS mark 5
Got SDP version 2 and unique parts [- 8287664629104701326 IN IP4 127.0.0.1]
Found RTP audio format 111
Found audio description format opus for ID 111
Capabilities: us - (opus), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
> 0xb4c79598 – Strict RTP learning after remote address set to: 192.168.6.100:62748
Peer audio RTP is at port 192.168.6.100:62748
Looking for 100 in from-extensions (domain 192.168.4.189)
sip_route_dump: route/path hop: sip:06r3i1vi@192.0.2.123;transport=wss;ob

<— Transmitting (no NAT) to 192.168.6.100:5060 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.123;branch=z9hG4bK4796544;received=192.168.6.100
From: “MyApp” sip:User2@192.168.4.189;tag=vlk9qceijm
To: sip:100@192.168.4.189
Call-ID: 4dg3sdggf4fnq0f0v8ua
CSeq: 1751 INVITE
Server: Asterisk PBX 16.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:100@192.168.4.189:5060;transport=ws
Content-Length: 0

<------------>
– Executing [100@from-extensions:1] Dial(“SIP/User2-0000001a”, “SIP/User1,30”) in new stack
== DTLS ECDH initialized (automatic), faster PFS enabled
== Using SIP RTP CoS mark 5
Audio is at 15232
Adding codec opus to SDP
Reliably Transmitting (no NAT) to 192.168.6.100:60303:
INVITE sip:nu9vf5ch@192.0.2.130;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.4.189:5060;branch=z9hG4bK3d523880
Max-Forwards: 70
From: “User 2” sip:200@192.168.4.189;tag=as7375dc3c
To: sip:nu9vf5ch@192.0.2.130;transport=wss
Contact: sip:200@192.168.4.189:5060;transport=ws
Call-ID: 2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 16.20.0
Date: Wed, 15 Dec 2021 12:10:54 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 619

v=0
o=root 93354871 93354871 IN IP4 192.168.4.189
s=Asterisk PBX 16.20.0
c=IN IP4 192.168.4.189
t=0 0
m=audio 15232 UDP/TLS/RTP/SAVPF 107
a=rtpmap:107 opus/48000/2
a=fmtp:107 useinbandfec=1
a=maxptime:20
a=ice-ufrag:343c15e9445fa2c704998700074c7a01
a=ice-pwd:32df3fce6509073c2bae919f409b24d1
a=candidate:Hc0a804bd 1 UDP 2130706431 192.168.4.189 15232 typ host
a=candidate:Hc0a804bd 2 UDP 2130706430 192.168.4.189 15233 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 6A:4B:E1:73:69:6E:92:99:EE:55:FA:18:11:E4:AB:D2:EB:EE:8C:65:05:26:4A:56:AD:D5:14:C3:2A:EA:38:12
a=rtcp-mux
a=sendrecv

-- Called SIP/User1

<— SIP read from WS:192.168.6.100:60303 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WS 192.168.4.189:5060;branch=z9hG4bK3d523880
To: sip:nu9vf5ch@192.0.2.130;transport=wss
From: “User 2” sip:200@192.168.4.189;tag=as7375dc3c
Call-ID: 2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060
CSeq: 102 INVITE
Supported: outbound
User-Agent: MyApp App
Content-Length: 0

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

<— SIP read from WS:192.168.6.100:60303 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 192.168.4.189:5060;branch=z9hG4bK3d523880
To: sip:nu9vf5ch@192.0.2.130;transport=wss;tag=1sidihqpb9
From: “User 2” sip:200@192.168.4.189;tag=as7375dc3c
Call-ID: 2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060
CSeq: 102 INVITE
Contact: sip:nu9vf5ch@192.0.2.130;transport=wss
Supported: outbound
User-Agent: MyApp App
Content-Length: 0

<------------->
— (10 headers 0 lines) —
sip_route_dump: route/path hop: sip:nu9vf5ch@192.0.2.130;transport=wss
– SIP/User1-0000001b is ringing

<— Transmitting (no NAT) to 192.168.6.100:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 192.0.2.123;branch=z9hG4bK4796544;received=192.168.6.100
From: “MyApp” sip:User2@192.168.4.189;tag=vlk9qceijm
To: sip:100@192.168.4.189;tag=as4cb29512
Call-ID: 4dg3sdggf4fnq0f0v8ua
CSeq: 1751 INVITE
Server: Asterisk PBX 16.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:100@192.168.4.189:5060;transport=ws
Content-Length: 0

<------------>
> 0x9d85c18 – Strict RTP learning after remote address set to: 192.168.6.100:54342

<— SIP read from WS:192.168.6.100:60303 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.4.189:5060;branch=z9hG4bK3d523880
To: sip:nu9vf5ch@192.0.2.130;transport=wss;tag=1sidihqpb9
From: “User 2” sip:200@192.168.4.189;tag=as7375dc3c
Call-ID: 2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060
CSeq: 102 INVITE
Contact: sip:nu9vf5ch@192.0.2.130;transport=wss
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: MyApp App
Content-Type: application/sdp
Content-Length: 2183

v=0
o=- 9074019485158578614 2 IN IP4 127.0.0.1
s=-
t=0 0
a=msid-semantic: WMS afpSNDcG6f5y9MmLUNJwOBJtSxegd8UufUqZ
m=audio 54342 UDP/TLS/RTP/SAVPF 107
c=IN IP4 192.168.6.100
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:3110093121 1 udp 2122260223 192.168.6.100 54342 typ host generation 0 network-id 5 network-cost 50
a=candidate:633891456 1 udp 2122189567 2001::348b:fb58:1401:1a78:62d2:af60 54343 typ host generation 0 network-id 4 network-cost 50
a=candidate:4027572201 1 udp 2122131711 2409:4071:208d:e6fd:952a:2959:97f3:b9d9 54344 typ host generation 0 network-id 2 network-cost 10
a=candidate:951146822 1 udp 2122066175 2409:4071:208d:e6fd:a83e:c7e2:fb68:a2e9 54345 typ host generation 0 network-id 3 network-cost 10
a=candidate:419066100 1 udp 2121998079 192.168.225.28 54346 typ host generation 0 network-id 1 network-cost 10
a=candidate:4158897585 1 tcp 1518280447 192.168.6.100 9 typ host tcptype active generation 0 network-id 5 network-cost 50
a=candidate:1800115824 1 tcp 1518209791 2001::348b:fb58:1401:1a78:62d2:af60 9 typ host tcptype active generation 0 network-id 4 network-cost 50
a=candidate:3196855065 1 tcp 1518151935 2409:4071:208d:e6fd:952a:2959:97f3:b9d9 9 typ host tcptype active generation 0 network-id 2 network-cost 10
a=candidate:1983030710 1 tcp 1518086399 2409:4071:208d:e6fd:a83e:c7e2:fb68:a2e9 9 typ host tcptype active generation 0 network-id 3 network-cost 10
a=candidate:1450806276 1 tcp 1518018303 192.168.225.28 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:s8vd
a=ice-pwd:rnSiqKBJBeYqszvdLULeGtQr
a=ice-options:trickle
a=fingerprint:sha-256 4B:FC:C6:39:4D:B7:7F:F9:C4:13:17:BB:36:F4:DC:E2:23:6E:B8:34:C4:91:FF:59:A5:D2:C3:23:EA:60:66:0D
a=setup:active
a=mid:0
a=sendrecv
a=msid:afpSNDcG6f5y9MmLUNJwOBJtSxegd8UufUqZ af89b385-5f7b-458e-84a2-b1bcb8ae2d8c
a=rtcp-mux
a=rtpmap:107 opus/48000/2
a=fmtp:107 minptime=10;useinbandfec=1
a=ssrc:568529376 cname:E08vXD1xtBslAi4T
a=ssrc:568529376 msid:afpSNDcG6f5y9MmLUNJwOBJtSxegd8UufUqZ af89b385-5f7b-458e-84a2-b1bcb8ae2d8c
a=ssrc:568529376 mslabel:afpSNDcG6f5y9MmLUNJwOBJtSxegd8UufUqZ
a=ssrc:568529376 label:af89b385-5f7b-458e-84a2-b1bcb8ae2d8c
<------------->
— (12 headers 33 lines) —
Got SDP version 2 and unique parts [- 9074019485158578614 IN IP4 127.0.0.1]
Found RTP audio format 107
Found audio description format opus for ID 107
Capabilities: us - (opus), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus)
Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing)
Peer audio RTP is at port 192.168.6.100:54342
sip_route_dump: route/path hop: sip:nu9vf5ch@192.0.2.130;transport=wss
set_destination: Parsing sip:nu9vf5ch@192.0.2.130;transport=wss for address/port to send to
set_destination: URI is for WebSocket, we can’t set destination
Transmitting (no NAT) to 192.0.2.130:5060:
ACK sip:nu9vf5ch@192.0.2.130;transport=wss SIP/2.0
Via: SIP/2.0/WS 192.168.4.189:5060;branch=z9hG4bK23b696a7
Max-Forwards: 70
From: “User 2” sip:200@192.168.4.189;tag=as7375dc3c
To: sip:nu9vf5ch@192.0.2.130;transport=wss;tag=1sidihqpb9
Contact: sip:200@192.168.4.189:5060;transport=ws
Call-ID: 2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 16.20.0
Content-Length: 0

-- SIP/User1-0000001b answered SIP/User2-0000001a

Audio is at 19532
Adding codec opus to SDP

<— Reliably Transmitting (no NAT) to 192.168.6.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.123;branch=z9hG4bK4796544;received=192.168.6.100
From: “MyApp” sip:User2@192.168.4.189;tag=vlk9qceijm
To: sip:100@192.168.4.189;tag=as4cb29512
Call-ID: 4dg3sdggf4fnq0f0v8ua
CSeq: 1751 INVITE
Server: Asterisk PBX 16.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:100@192.168.4.189:5060;transport=ws
Content-Type: application/sdp
Content-Length: 620

v=0
o=root 860846233 860846233 IN IP4 192.168.4.189
s=Asterisk PBX 16.20.0
c=IN IP4 192.168.4.189
t=0 0
m=audio 19532 UDP/TLS/RTP/SAVPF 111
a=rtpmap:111 opus/48000/2
a=fmtp:111 useinbandfec=1
a=maxptime:20
a=ice-ufrag:7d162f4008b22c0e223d2cdc67b5f39d
a=ice-pwd:2290c4285aef1c3b489cb4b67562302f
a=candidate:Hc0a804bd 1 UDP 2130706431 192.168.4.189 19532 typ host
a=candidate:Hc0a804bd 2 UDP 2130706430 192.168.4.189 19533 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 6A:4B:E1:73:69:6E:92:99:EE:55:FA:18:11:E4:AB:D2:EB:EE:8C:65:05:26:4A:56:AD:D5:14:C3:2A:EA:38:12
a=rtcp-mux
a=sendrecv

<------------>
– Channel SIP/User1-0000001b joined ‘simple_bridge’ basic-bridge <72562c82-cd01-4781-bf88-7a76c76a0f5a>
– Channel SIP/User2-0000001a joined ‘simple_bridge’ basic-bridge <72562c82-cd01-4781-bf88-7a76c76a0f5a>
> 0x9d85c18 – Strict RTP learning after ICE completion
> 0x9d85c18 – Strict RTP learning after remote address set to: 192.168.6.100:54342

<— SIP read from WS:192.168.6.100:60326 —>
ACK sip:100@192.168.4.189:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.123;branch=z9hG4bK9627095
Max-Forwards: 70
To: sip:100@192.168.4.189;tag=as4cb29512
From: “MyApp” sip:User2@192.168.4.189;tag=vlk9qceijm
Call-ID: 4dg3sdggf4fnq0f0v8ua
CSeq: 1751 ACK
Supported: outbound
User-Agent: MyApp App
Content-Length: 0

<------------->
— (10 headers 0 lines) —
> 0xb4c79598 – Strict RTP learning after ICE completion
> 0xb4c79598 – Strict RTP learning after remote address set to: 192.168.6.100:62748
> 0x9d85c18 – Strict RTP switching to RTP target address 192.168.6.100:54342 as source
> 0xb4c79598 – Strict RTP switching to RTP target address 192.168.6.100:62748 as source
[2021-12-15 17:41:00] ERROR[4669]: iostream.c:552 ast_iostream_close: SSL_shutdown() failed: error:00000005:lib(0):func(0):DH lib, Underlying BIO error: Bad file descriptor
== WebSocket connection from ‘192.168.6.100:55375’ forcefully closed due to fatal write error

<— SIP read from WS:192.168.6.100:60303 —>
BYE sip:200@192.168.4.189:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.130;branch=z9hG4bK544138
Max-Forwards: 70
To: sip:200@192.168.4.189;tag=as7375dc3c
From: “MyApp” sip:nu9vf5ch@192.0.2.130;transport=wss;tag=1sidihqpb9
Call-ID: 2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060
CSeq: 103 BYE
Supported: outbound
User-Agent: MyApp App
Content-Length: 0

<------------->
— (10 headers 0 lines) —
Scheduling destruction of SIP dialog ‘2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 192.168.6.100:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.130;branch=z9hG4bK544138;received=192.168.6.100
From: “MyApp” sip:nu9vf5ch@192.0.2.130;transport=wss;tag=1sidihqpb9
To: sip:200@192.168.4.189;tag=as7375dc3c
Call-ID: 2a7cfcf35c92042746648fd25de965df@192.168.4.189:5060
CSeq: 103 BYE
Server: Asterisk PBX 16.20.0
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/User1-0000001b left ‘simple_bridge’ basic-bridge <72562c82-cd01-4781-bf88-7a76c76a0f5a>
– Channel SIP/User2-0000001a left ‘simple_bridge’ basic-bridge <72562c82-cd01-4781-bf88-7a76c76a0f5a>
== Spawn extension (from-extensions, 100, 1) exited non-zero on ‘SIP/User2-0000001a’
Scheduling destruction of SIP dialog ‘4dg3sdggf4fnq0f0v8ua’ in 32000 ms (Method: ACK)
set_destination: Parsing sip:06r3i1vi@192.0.2.123;transport=wss;ob for address/port to send to
set_destination: URI is for WebSocket, we can’t set destination
Reliably Transmitting (no NAT) to 192.168.6.100:5060:
BYE sip:06r3i1vi@192.0.2.123;transport=wss;ob SIP/2.0
Via: SIP/2.0/WS 192.168.4.189:5060;branch=z9hG4bK32942600
Max-Forwards: 70
From: sip:100@192.168.4.189;tag=as4cb29512
To: “MyApp” sip:User2@192.168.4.189;tag=vlk9qceijm
Call-ID: 4dg3sdggf4fnq0f0v8ua
CSeq: 102 BYE
User-Agent: Asterisk PBX 16.20.0
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from WS:192.168.6.100:60326 —>
SIP/2.0 200 OK
Via: SIP/2.0/WS 192.168.4.189:5060;branch=z9hG4bK32942600
To: “MyApp” sip:User2@192.168.4.189;tag=vlk9qceijm
From: sip:100@192.168.4.189;tag=as4cb29512
Call-ID: 4dg3sdggf4fnq0f0v8ua
CSeq: 102 BYE
Supported: outbound
User-Agent: MyApp App
Content-Length: 0

<------------->
— (9 headers 0 lines) —
SIP Response message for INCOMING dialog BYE arrived
Really destroying SIP dialog ‘4dg3sdggf4fnq0f0v8ua’ Method: ACK
Really destroying SIP dialog ‘4dg3sn46tks0s861b7mk’ Method: BYE

Thanks

2

Direct media is not supported when ICE or SRTP is in use, such as when WebRTC is used.

3

Thanks jcolp for your prompt reply.

I have already disabled iceServers in my app. You mean I should remove ice reference from munged SDP and in Asterisk settings also? I will try to remove iceTransportPolicy and extmapAllowMixed also.from my app.

Below is my getUserMediaRequest

https://192.168.4.189:4443/static/index.html, { iceServers: , iceTransportPolicy: all, bundlePolicy: balanced, rtcpMuxPolicy: require, iceCandidatePoolSize: 0, sdpSemantics: “unified-plan”, extmapAllowMixed: true }

Moreover I believe WebRTC require SRTP to work. Is it mandatory to use RTP only for directmedia?

Best Regards
Manish

4

Yes, it is mandatory to use RTP for direct media. It will not work with WebRTC.

5

Thanks jcolp for your reply.

One more question, By any chance directrtpsetup will work, or do you suggest any alternate way for having P2P call using WebRTC?

I really need to have P2P call using WebRTC using Asterisk

Best Regards
Manish

6

No, there is no way. Direct RTP setup does not support such a thing either.

7

Thanks jcolp for your reply.

8

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