rtp.conf
rtpstart=10000
rtpend=20000
icesupport=yes
stunaddr=stun.l.google.com:19302
sip.conf
externip=167.235.238.222
<--- SIP read from WS:103.184.238.161:4577 --->
INVITE sip:300@thevoizbox.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.41;branch=z9hG4bK2661406
To: <sip:300@thevoizbox.com>
From: "Demo" <sip:787@thevoizbox.com>;tag=sq2hh0bju8
CSeq: 1 INVITE
Call-ID: 71lvq8fknmpgun8ugmvv
Max-Forwards: 70
Contact: <sip:175l9m3k@192.0.2.41;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Thevoizbox Webrtc Phone
Content-Type: application/sdp
Content-Length: 2581
v=0
o=- 7597307579517885970 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS b7a02a1d-b853-45e8-a740-b2ad4f77f461
m=audio 4600 UDP/TLS/RTP/SAVPF 111 63 103 9 0 8 105 13 110 113 126
c=IN IP4 103.184.238.161
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1945132609 1 udp 2122260223 192.168.31.68 49323 typ host generation 0 network-id 4 network-cost 10
a=candidate:3268480406 1 udp 2122197247 2409:4033:4e1a:757f:7d1d:1107:105c:61e4 41506 typ host generation 0 network-id 1 network-cost 900
a=candidate:2562614748 1 udp 2122131711 2409:4073:4d8b:1d58:9fa9:3e1:5d02:31fb 42369 typ host generation 0 network-id 2 network-cost 900
a=candidate:4165055168 1 udp 2122063615 25.235.82.233 40637 typ host generation 0 network-id 3 network-cost 900
a=candidate:2717703118 1 udp 1686052607 103.184.238.161 4600 typ srflx raddr 192.168.31.68 rport 49323 generation 0 network-id 4 network-cost 10
a=candidate:2371521237 1 tcp 1518280447 192.168.31.68 9 typ host tcptype active generation 0 network-id 4 network-cost 10
a=candidate:1014749442 1 tcp 1518217471 2409:4033:4e1a:757f:7d1d:1107:105c:61e4 9 typ host tcptype active generation 0 network-id 1 network-cost 900
a=candidate:1712632136 1 tcp 1518151935 2409:4073:4d8b:1d58:9fa9:3e1:5d02:31fb 9 typ host tcptype active generation 0 network-id 2 network-cost 900
a=candidate:116090452 1 tcp 1518083839 25.235.82.233 9 typ host tcptype active generation 0 network-id 3 network-cost 900
a=ice-ufrag:lHZS
a=ice-pwd:Gyq7So7bDVD8UNlppRugZiZi
a=ice-options:trickle
a=fingerprint:sha-256 9E:29:E4:41:46:2D:D1:F6:23:47:2A:09:78:B0:4E:4A:EF:CE:7D:71:AD:68:1E:E6:E2:0A:9B:3E:C5:2E:22:B8
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:b7a02a1d-b853-45e8-a740-b2ad4f77f461 b003b490-7d95-4918-a569-2b48bf3c2263
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:103 ISAC/16000
a=rtpmap:9 G722/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:110 telephone-event/48000
a=rtpmap:113 telephone-event/16000
a=rtpmap:126 telephone-event/8000
a=ssrc:2222438991 cname:z1hmwZf5xqFvNlRD
a=ssrc:2222438991 msid:b7a02a1d-b853-45e8-a740-b2ad4f77f461 b003b490-7d95-4918-a569-2b48bf3c2263
<------------->
--- (13 headers 48 lines) ---
Using INVITE request as basis request - 71lvq8fknmpgun8ugmvv
Found peer '787' for '787' from 103.184.238.161:4577
== Using SIP RTP CoS mark 5
Got SDP version 2 and unique parts [- 7597307579517885970 IN IP4 127.0.0.1]
Found RTP audio format 111
Found RTP audio format 63
Found RTP audio format 103
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 105
Found RTP audio format 13
Found RTP audio format 110
Found RTP audio format 113
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format red for ID 63
Found unknown media description format ISAC for ID 103
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found unknown media description format CN for ID 105
Found audio description format CN for ID 13
Found unknown media description format telephone-event for ID 110
Found unknown media description format telephone-event for ID 113
Found audio description format telephone-event for ID 126
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw|g722|opus)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x3 (telephone-event|CN|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 103.184.238.161:4600
Looking for 300 in internal-webrtc (domain thevoizbox.com)
sip_route_dump: route/path hop: <sip:175l9m3k@192.0.2.41;transport=wss;ob>
<--- Transmitting (NAT) to 103.184.238.161:4577 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.41;branch=z9hG4bK2661406;received=103.184.238.161;rport=4577
From: "Demo" <sip:787@thevoizbox.com>;tag=sq2hh0bju8
To: <sip:300@thevoizbox.com>
Call-ID: 71lvq8fknmpgun8ugmvv
CSeq: 1 INVITE
Server: Thevoizbox PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:300@167.235.238.222:3250;transport=ws>
Content-Length: 0
<------------>
-- Executing [300@internal-webrtc:1] Set("SIP/787-00000005", "CALLERID(dnid)=787") in new stack
-- Executing [300@internal-webrtc:2] Set("SIP/787-00000005", "CALLERID(dnid)=787") in new stack
-- Executing [300@internal-webrtc:3] NoOp("SIP/787-00000005", "3") in new stack
-- Executing [300@internal-webrtc:4] NoOp("SIP/787-00000005", "787") in new stack
-- Executing [300@internal-webrtc:5] NoOp("SIP/787-00000005", "300") in new stack
-- Executing [300@internal-webrtc:6] Dial("SIP/787-00000005", "SIP/300") in new stack
== Using SIP RTP CoS mark 5
Audio is at 16208
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 103.184.238.161:2271:
INVITE sip:pcnt7qfa@192.0.2.216;transport=wss SIP/2.0
Via: SIP/2.0/WS 167.235.238.222:3250;branch=z9hG4bK4e3ee7e4;rport
Max-Forwards: 70
From: "Demo" <sip:787@167.235.238.222:3250>;tag=as29c70953
To: <sip:pcnt7qfa@192.0.2.216;transport=wss>
Contact: <sip:787@167.235.238.222:3250;transport=ws>
Call-ID: 52f1a98051f6aa6550811b7932a643f3@167.235.238.222:3250
CSeq: 102 INVITE
User-Agent: Thevoizbox PBX
Date: Mon, 28 Nov 2022 18:20:35 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Remote-Party-ID: "Demo" <sip:787@167.235.238.222>;party=calling;privacy=off;screen=no
Content-Type: application/sdp
Content-Length: 712
v=0
o=root 679046439 679046439 IN IP4 167.235.238.222
s=Asterisk PBX certified/18.9-cert2
c=IN IP4 167.235.238.222
t=0 0
m=audio 16208 UDP/TLS/RTP/SAVPF 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=ice-ufrag:7a1bbca4452e6f573f3ec3193cb91496
a=ice-pwd:7453ee6f7fc06e260d0613493c0722d7
a=candidate:Ha7ebeede 1 UDP 2130706431 167.235.238.222 16208 typ host
a=candidate:Ha7ebeede 2 UDP 2130706430 167.235.238.222 16209 typ host
a=connection:new
a=setup:actpass
a=fingerprint:SHA-256 9F:F0:BD:C9:10:B3:86:32:FA:AB:B0:B6:F8:1E:16:FC:6D:34:79:EB:94:AB:97:E7:C8:2F:59:3C:10:94:C4:22
a=rtcp-mux
a=sendrecv
---
-- Called SIP/300
<--- SIP read from WS:103.184.238.161:2271 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WS 167.235.238.222:3250;branch=z9hG4bK4e3ee7e4;rport
From: "Demo" <sip:787@167.235.238.222:3250>;tag=as29c70953
To: <sip:pcnt7qfa@192.0.2.216;transport=wss>
CSeq: 102 INVITE
Call-ID: 52f1a98051f6aa6550811b7932a643f3@167.235.238.222:3250
Supported: outbound
User-Agent: Thevoizbox Webrtc Phone
Content-Length: 0
<------------->
--- (9 headers 0 lines) ---
<--- SIP read from WS:103.184.238.161:2271 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WS 167.235.238.222:3250;branch=z9hG4bK4e3ee7e4;rport
From: "Demo" <sip:787@167.235.238.222:3250>;tag=as29c70953
To: <sip:pcnt7qfa@192.0.2.216;transport=wss>;tag=abl9dnbkbk
CSeq: 102 INVITE
Call-ID: 52f1a98051f6aa6550811b7932a643f3@167.235.238.222:3250
Supported: outbound
User-Agent: Thevoizbox Webrtc Phone
Contact: <sip:pcnt7qfa@192.0.2.216;transport=wss>
Content-Length: 0
<------------->
--- (10 headers 0 lines) ---
sip_route_dump: route/path hop: <sip:pcnt7qfa@192.0.2.216;transport=wss>
-- SIP/300-00000006 is ringing
<--- Transmitting (NAT) to 103.184.238.161:4577 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 192.0.2.41;branch=z9hG4bK2661406;received=103.184.238.161;rport=4577
From: "Demo" <sip:787@thevoizbox.com>;tag=sq2hh0bju8
To: <sip:300@thevoizbox.com>;tag=as156e5e46
Call-ID: 71lvq8fknmpgun8ugmvv
CSeq: 1 INVITE
Server: Thevoizbox PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:300@167.235.238.222:3250;transport=ws>
Content-Length: 0
<------------>
<--- SIP read from WS:103.184.238.161:2271 --->
SIP/2.0 200 OK
Via: SIP/2.0/WS 167.235.238.222:3250;branch=z9hG4bK4e3ee7e4;rport
From: "Demo" <sip:787@167.235.238.222:3250>;tag=as29c70953
To: <sip:pcnt7qfa@192.0.2.216;transport=wss>;tag=abl9dnbkbk
CSeq: 102 INVITE
Call-ID: 52f1a98051f6aa6550811b7932a643f3@167.235.238.222:3250
Supported: outbound
User-Agent: Thevoizbox Webrtc Phone
Allow: ACK,BYE,CANCEL,INFO,INVITE,MESSAGE,NOTIFY,OPTIONS,PRACK,REFER,REGISTER,SUBSCRIBE
Contact: <sip:pcnt7qfa@192.0.2.216;transport=wss>
Content-Type: application/sdp
Content-Length: 1052
v=0
o=mozilla...THIS_IS_SDPARTA-99.0 3010539372216082334 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 C1:9F:D2:12:C0:FF:64:B6:72:DC:8A:C7:71:A5:04:EA:7A:5B:D9:F4:4C:78:3D:E9:17:A0:88:81:E6:2D:35:10
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 4604 UDP/TLS/RTP/SAVPF 0 8 101
c=IN IP4 103.184.238.161
a=candidate:0 1 UDP 2122187007 192.168.31.177 41996 typ host
a=candidate:2 1 UDP 2122252543 192.168.1.10 35810 typ host
a=candidate:4 1 TCP 2105458943 192.168.31.177 9 typ host tcptype active
a=candidate:5 1 TCP 2105524479 192.168.1.10 9 typ host tcptype active
a=candidate:3 1 UDP 1686052863 103.184.238.161 4604 typ srflx raddr 192.168.1.10 rport 35810
a=sendrecv
a=fmtp:101 0-15
a=ice-pwd:9528580fd2c4efa89c01258279a08421
a=ice-ufrag:6f1e45d9
a=mid:0
a=msid:{965265cb-a092-4a44-a29f-791e28c1a230} {8d4d1c1d-0a3f-4e01-a41f-38897152d02a}
a=rtcp-mux
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:active
a=ssrc:3134691940 cname:{a54af554-4887-4853-800e-5fe128aca2fa}
<------------->
--- (12 headers 27 lines) ---
Got SDP version 0 and unique parts [mozilla...THIS_IS_SDPARTA-99.0 3010539372216082334 IN IP4 0.0.0.0]
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw|alaw)/video=(nothing)/text=(nothing), combined - (ulaw|alaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
Peer audio RTP is at port 103.184.238.161:4604
sip_route_dump: route/path hop: <sip:pcnt7qfa@192.0.2.216;transport=wss>
Transmitting (NAT) to 103.184.238.161:2271:
ACK sip:pcnt7qfa@192.0.2.216;transport=wss SIP/2.0
Via: SIP/2.0/WS 167.235.238.222:3250;branch=z9hG4bK3f3ae92a;rport
Max-Forwards: 70
From: "Demo" <sip:787@167.235.238.222:3250>;tag=as29c70953
To: <sip:pcnt7qfa@192.0.2.216;transport=wss>;tag=abl9dnbkbk
Contact: <sip:787@167.235.238.222:3250;transport=ws>
Call-ID: 52f1a98051f6aa6550811b7932a643f3@167.235.238.222:3250
CSeq: 102 ACK
User-Agent: Thevoizbox PBX
Content-Length: 0
---
-- SIP/300-00000006 answered SIP/787-00000005
Audio is at 13856
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (NAT) to 103.184.238.161:4577 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.41;branch=z9hG4bK2661406;received=103.184.238.161;rport=4577
From: "Demo" <sip:787@thevoizbox.com>;tag=sq2hh0bju8
To: <sip:300@thevoizbox.com>;tag=as156e5e46
Call-ID: 71lvq8fknmpgun8ugmvv
CSeq: 1 INVITE
Server: Thevoizbox PBX
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: <sip:300@167.235.238.222:3250;transport=ws>
Content-Type: application/sdp
Content-Length: 688
v=0
o=root 310467301 310467301 IN IP4 167.235.238.222
s=Asterisk PBX certified/18.9-cert2
c=IN IP4 167.235.238.222
t=0 0
m=audio 13856 UDP/TLS/RTP/SAVPF 0 8 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:150
a=ice-ufrag:4acb69df4326d16466674057526ec9d0
a=ice-pwd:400175506d023553563a8f0817ee69fe
a=candidate:Ha7ebeede 1 UDP 2130706431 167.235.238.222 13856 typ host
a=candidate:Ha7ebeede 2 UDP 2130706430 167.235.238.222 13857 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9F:F0:BD:C9:10:B3:86:32:FA:AB:B0:B6:F8:1E:16:FC:6D:34:79:EB:94:AB:97:E7:C8:2F:59:3C:10:94:C4:22
a=rtcp-mux
a=sendrecv
<------------>
-- Channel SIP/300-00000006 joined 'simple_bridge' basic-bridge <c38e4d46-df0c-444f-9405-0d4214c906ff>
-- Channel SIP/787-00000005 joined 'simple_bridge' basic-bridge <c38e4d46-df0c-444f-9405-0d4214c906ff>
<--- SIP read from WS:103.184.238.161:4577 --->
ACK sip:300@167.235.238.222:3250;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.41;branch=z9hG4bK8695152
To: <sip:300@thevoizbox.com>;tag=as156e5e46
From: "Demo" <sip:787@thevoizbox.com>;tag=sq2hh0bju8
CSeq: 1 ACK
Call-ID: 71lvq8fknmpgun8ugmvv
Max-Forwards: 70
Supported: outbound
User-Agent: Thevoizbox Webrtc Phone
Content-Length: 0
<------------->