I tried to configure Asterisk 18 in docker with WebRTC and SIP
I have peers - 501, 502 - WebRTC (Chrome Browser), 1003 - Zoiper
10.7.97.171 - docker host IP
10.5.152.12 - clients IP (501 502 1003)
172.17.0.27 - asterisk in docker
rtp.conf
[general]
rtpstart=10050
rtpend=10100
icesupport=true
stunaddr=stun.l.google.com:19302
http.conf
text[general]
enabled=yes
bindaddr=0.0.0.0
bindport=8088
tlsenable=yes
tlsbindaddr=0.0.0.0:8089
tlscertfile=/etc/asterisk/keys/my_domain-peer.pem
tlsprivatekey=/etc/asterisk/keysmy_domain-key.pem
pjsip.conf
[transport-jigasi]
type = transport
protocol = udp
bind = 0.0.0.0:5160
external_media_address=10.7.97.171
external_signaling_address=10.7.97.171
[jigasi_auth]
type = auth
auth_type = userpass
password = dddddddddddddd
username = jigasi
[jigasi]
type = aor
max_contacts = 1
remove_existing = yes
[jigasi]
type = endpoint
transport = transport-jigasi
context = public
direct_media = no
disallow = all
allow = ulaw
allow = speex
auth = jigasi_auth
aors = jigasi
rtp_symmetric = yes
force_rport = yes
rewrite_contact = yes
[1003]
type = auth
username = 1003
password = 1234
[1003]
type = endpoint
transport = transport-jigasi
context = public
dtmf_mode=rfc4733
auth = 1003
outbound_auth = 1003
aors = 1003
disallow = all
allow = g722
allow = speex
allow = alaw
[1004]
type = aor
max_contacts = 1
[502]
type=aor
max_contacts=5
remove_existing=yes
[502]
type=auth
auth_type=userpass
username=502
password=webrtc_client ;
[502]
type=endpoint
dtmf_mode=rfc4733
aors=502
auth=502
dtls_auto_generate_cert=yes
webrtc=yes
direct_media = no
; Setting webrtc=yes is a shortcut for setting the following options:
; use_avpf=yes
; media_encryption=dtls
; dtls_verify=fingerprint
; dtls_setup=actpass
; ice_support=yes
; media_use_received_transport=yes
; rtcp_mux=yes
context=public
disallow=all
allow=g722, speex,opus,ulaw
[transport-wss]
type=transport
protocol=wss
bind=0.0.0.0
extensions.conf
[public]
exten => 1003,1,Dial(PJSIP/1003,10)
exten => jigasi,1,Dial(PJSIP/jigasi,10)
exten => 501,1,Dial(PJSIP/501,10)
exten => 502,1,Dial(PJSIP/502,10)
When I call from 502 (webrtc) to 1003 it works fine and the sound goes both ways
success call from 502 to 1003
-- Added contact 'sip:9qlp9nhl@10.5.152.12:41044;transport=ws;x-ast-orig-host=jqtvs5k9ie78.invalid:0' to AOR '502' with expiration of 600 seconds
<— Transmitting SIP response (496 bytes) to WSS:10.5.152.12:41044 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;rport=41044;received=10.5.152.12;branch=z9hG4bK1737204
Call-ID: 0reku46bg2kn1nmlfpd3
From: sip:502@my.domain.net;tag=rg6gd4phqj
To: sip:502@my.domain.net;tag=z9hG4bK1737204
CSeq: 3 REGISTER
Date: Mon, 18 Mar 2024 20:15:19 GMT
Contact: sip:mn7stugv@qiu4it0jrnv1.invalid;transport=ws;expires=118
Contact: sip:9qlp9nhl@jqtvs5k9ie78.invalid;transport=ws;expires=599
Server: Asterisk PBX 18.20.2
Content-Length: 0
<— Received SIP request (343 bytes) from WSS:10.5.152.12:41044 —>
OPTIONS sip:502@my.domain.net:8089 SIP/2.0
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;branch=z9hG4bK2404885
To: sip:502@my.domain.net:8089
From: sip:502@my.domain.net:8089;tag=2vbp0qmvkp
CSeq: 1 OPTIONS
Call-ID: 0rekuqsk9kpar0pkve1f
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.20.1
Content-Length: 0
<— Transmitting SIP response (801 bytes) to WSS:10.5.152.12:41044 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;rport=41044;received=10.5.152.12;branch=z9hG4bK4746306
Call-ID: 0reku6ivo6rcmtibas0n
From: sip:502@my.domain.net;tag=3vssp2ku5g
To: sip:502@my.domain.net;tag=z9hG4bK4746306
CSeq: 2 OPTIONS
Accept: application/xpidf+xml, application/cpim-pidf+xml, application/dialog-info+xml, application/pidf+xml, application/simple-message-summary, application/pidf+xml, application/dialog-info+xml, application/simple-message-summary, application/sdp, message/sipfrag;version=2.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub
Accept-Encoding: identity
Accept-Language: en
Server: Asterisk PBX 18.20.2
Content-Length: 0
<— Received SIP request (2178 bytes) from WSS:10.5.152.12:41044 —>
INVITE sip:1003@my.domain.net:8089 SIP/2.0
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;branch=z9hG4bK8569743
To: sip:1003@my.domain.net:8089
From: sip:502@my.domain.net:8089;tag=cgttupilb1
CSeq: 1 INVITE
Call-ID: 0reku7oc329lkhvr86to
Max-Forwards: 70
Contact: sip:9qlp9nhl@jqtvs5k9ie78.invalid;transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.20.1
Content-Type: application/sdp
Content-Length: 1675
v=0
o=- 8713593786049119125 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 41c57e10-e226-4aa0-96ce-41d897286727
m=audio 55546 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 172.23.0.1
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4259692166 1 udp 2122260223 172.23.0.1 55546 typ host generation 0 network-id 1
a=candidate:2746971354 1 udp 2122194687 10.5.152.12 43699 typ host generation 0 network-id 2
a=candidate:1580867998 1 udp 2122129151 172.17.0.1 52294 typ host generation 0 network-id 3
a=candidate:3875661938 1 udp 2122063615 192.168.1.96 35092 typ host generation 0 network-id 4 network-cost 10
a=ice-ufrag:b+V5
a=ice-pwd:UYZVQd//QW1KA9/dozqeUzTl
a=ice-options:trickle
a=fingerprint:sha-256 D3:5A:E3:70:52:8C:8F:2F:00:79:A4:FA:FC:FE:44:28:84:F2:2A:ED:83:87:FA:90:5B:67:84:02:1F:40:9D:24
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
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:41c57e10-e226-4aa0-96ce-41d897286727 2ca441cc-d23b-4640-b1c8-e3572e69beb2
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:40419412 cname:FOs0P3mO9L/yoRRO
a=ssrc:40419412 msid:41c57e10-e226-4aa0-96ce-41d897286727 2ca441cc-d23b-4640-b1c8-e3572e69beb2
<— Transmitting SIP response (472 bytes) to WSS:10.5.152.12:41044 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;rport=41044;received=10.5.152.12;branch=z9hG4bK8569743
Call-ID: 0reku7oc329lkhvr86to
From: sip:502@my.domain.net;tag=cgttupilb1
To: sip:1003@my.domain.net;tag=z9hG4bK8569743
CSeq: 1 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1710792927/e95ca4608d9b1abc2818347ae7834a3b”,opaque=“0ba6150b3dac5941”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 18.20.2
Content-Length: 0
<— Received SIP request (303 bytes) from WSS:10.5.152.12:41044 —>
ACK sip:1003@my.domain.net:8089 SIP/2.0
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;branch=z9hG4bK8569743
To: sip:1003@my.domain.net;tag=z9hG4bK8569743
From: sip:502@my.domain.net:8089;tag=cgttupilb1
Call-ID: 0reku7oc329lkhvr86to
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<— Received SIP request (2461 bytes) from WSS:10.5.152.12:41044 —>
INVITE sip:1003@my.domain.net:8089 SIP/2.0
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;branch=z9hG4bK8666530
To: sip:1003@my.domain.net:8089
From: sip:502@my.domain.net:8089;tag=cgttupilb1
CSeq: 2 INVITE
Call-ID: 0reku7oc329lkhvr86to
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“502”, realm=“asterisk”, nonce=“1710792927/e95ca4608d9b1abc2818347ae7834a3b”, uri=“sip:1003@my.domain.net:8089”, response=“ae220a8d148b9079eb05b6a102a690af”, opaque=“0ba6150b3dac5941”, qop=auth, cnonce=“i4b0h77d2bum”, nc=00000001
Contact: sip:9qlp9nhl@jqtvs5k9ie78.invalid;transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.20.1
Content-Type: application/sdp
Content-Length: 1675
v=0
o=- 8713593786049119125 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 41c57e10-e226-4aa0-96ce-41d897286727
m=audio 55546 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 172.23.0.1
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:4259692166 1 udp 2122260223 172.23.0.1 55546 typ host generation 0 network-id 1
a=candidate:2746971354 1 udp 2122194687 10.5.152.12 43699 typ host generation 0 network-id 2
a=candidate:1580867998 1 udp 2122129151 172.17.0.1 52294 typ host generation 0 network-id 3
a=candidate:3875661938 1 udp 2122063615 192.168.1.96 35092 typ host generation 0 network-id 4 network-cost 10
a=ice-ufrag:b+V5
a=ice-pwd:UYZVQd//QW1KA9/dozqeUzTl
a=ice-options:trickle
a=fingerprint:sha-256 D3:5A:E3:70:52:8C:8F:2F:00:79:A4:FA:FC:FE:44:28:84:F2:2A:ED:83:87:FA:90:5B:67:84:02:1F:40:9D:24
a=setup:actpass
a=mid:0
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2 docs/native-code/rtp-hdrext/abs-send-time - src - Git at Google
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:41c57e10-e226-4aa0-96ce-41d897286727 2ca441cc-d23b-4640-b1c8-e3572e69beb2
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:40419412 cname:FOs0P3mO9L/yoRRO
a=ssrc:40419412 msid:41c57e10-e226-4aa0-96ce-41d897286727 2ca441cc-d23b-4640-b1c8-e3572e69beb2
<— Transmitting SIP response (301 bytes) to WSS:10.5.152.12:41044 —>
SIP/2.0 100 Trying
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;rport=41044;received=10.5.152.12;branch=z9hG4bK8666530
Call-ID: 0reku7oc329lkhvr86to
From: sip:502@my.domain.net;tag=cgttupilb1
To: sip:1003@my.domain.net
CSeq: 2 INVITE
Server: Asterisk PBX 18.20.2
Content-Length: 0
-- Executing [1003@public:1] Dial("PJSIP/502-00000038", "PJSIP/1003,10") in new stack
-- Called PJSIP/1003
<— Transmitting SIP request (999 bytes) to UDP:10.5.152.12:46083 —>
INVITE sip:1003@10.5.152.12:46083;transport=UDP;rinstance=812a4c0a0d178f44 SIP/2.0
Via: SIP/2.0/UDP 10.7.97.171:5160;rport;branch=z9hG4bKPjCtYn00Z8hUY3RrWN9yLkN6o8jhzU2F29
From: sip:502@172.17.0.27;tag=PoGkCT7sOu2oBhjEx9N1Cl1QwgCHOvNK
To: sip:1003@10.5.152.12;rinstance=812a4c0a0d178f44
Contact: sip:asterisk@10.7.97.171:5160
Call-ID: D6zKgsCme2P3zCoU5tOGj8mFP7Yf8hBb
CSeq: 29140 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.20.2
Content-Type: application/sdp
Content-Length: 287
v=0
o=- 1051440799 1051440799 IN IP4 10.7.97.171
s=Asterisk
c=IN IP4 10.7.97.171
t=0 0
m=audio 10068 RTP/AVP 9 110 8 101
a=rtpmap:9 G722/8000
a=rtpmap:110 speex/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:60
a=sendrecv
<— Received SIP response (320 bytes) from UDP:10.5.152.12:46083 —>
SIP/2.0 100 Trying
Via: SIP/2.0/UDP 10.7.97.171:5160;rport=5160;branch=z9hG4bKPjCtYn00Z8hUY3RrWN9yLkN6o8jhzU2F29
To: sip:1003@10.5.152.12;rinstance=812a4c0a0d178f44
From: sip:502@172.17.0.27;tag=PoGkCT7sOu2oBhjEx9N1Cl1QwgCHOvNK
Call-ID: D6zKgsCme2P3zCoU5tOGj8mFP7Yf8hBb
CSeq: 29140 INVITE
Content-Length: 0
<— Received SIP response (525 bytes) from UDP:10.5.152.12:46083 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 10.7.97.171:5160;rport=5160;branch=z9hG4bKPjCtYn00Z8hUY3RrWN9yLkN6o8jhzU2F29
Contact: sip:1003@10.5.152.12:46083
To: sip:1003@10.5.152.12;rinstance=812a4c0a0d178f44;tag=0feb8957
From: sip:502@172.17.0.27;tag=PoGkCT7sOu2oBhjEx9N1Cl1QwgCHOvNK
Call-ID: D6zKgsCme2P3zCoU5tOGj8mFP7Yf8hBb
CSeq: 29140 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
User-Agent: Z 5.5.13 v2.10.18.3
Allow-Events: presence, kpml, talk
Content-Length: 0
<— Received SIP response (933 bytes) from UDP:10.5.152.12:46083 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 10.7.97.171:5160;rport=5160;branch=z9hG4bKPjCtYn00Z8hUY3RrWN9yLkN6o8jhzU2F29
Require: timer
Contact: sip:1003@10.5.152.12:46083
To: sip:1003@10.5.152.12;rinstance=812a4c0a0d178f44;tag=0feb8957
From: sip:502@172.17.0.27;tag=PoGkCT7sOu2oBhjEx9N1Cl1QwgCHOvNK
Call-ID: D6zKgsCme2P3zCoU5tOGj8mFP7Yf8hBb
CSeq: 29140 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
User-Agent: Z 5.5.13 v2.10.18.3
Allow-Events: presence, kpml, talk
Content-Length: 327
v=0
o=Z 0 419883602 IN IP4 10.5.152.12
s=Z
c=IN IP4 10.5.152.12
t=0 0
m=audio 47670 RTP/AVP 9 106 0 8 3 101 98
a=rtpmap:106 opus/48000/2
a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=rtpmap:98 telephone-event/48000
a=fmtp:98 0-16
a=sendrecv
> 0x7f8084ede1b0 -- Strict RTP learning after remote address set to: 10.5.152.12:47670
> 0x7f8084ede1b0 -- Strict RTP switching to RTP target address 10.5.152.12:47670 as source
Got RTP packet from 10.5.152.12:47670 (type 95, seq 044387, ts 1826982922, len 000001)
<— Transmitting SIP request (398 bytes) to UDP:10.5.152.12:46083 —>
ACK sip:1003@10.5.152.12:46083 SIP/2.0
Via: SIP/2.0/UDP 10.7.97.171:5160;rport;branch=z9hG4bKPj2kcaWhr.5Jx0fbq6QlH2UoH446cE.Dms
From: sip:502@172.17.0.27;tag=PoGkCT7sOu2oBhjEx9N1Cl1QwgCHOvNK
To: sip:1003@10.5.152.12;rinstance=812a4c0a0d178f44;tag=0feb8957
Call-ID: D6zKgsCme2P3zCoU5tOGj8mFP7Yf8hBb
CSeq: 29140 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.20.2
Content-Length: 0
Got RTP packet from 10.5.152.12:47670 (type 09, seq 044388, ts 1826982922, len 000160)
– PJSIP/1003-00000039 answered PJSIP/502-00000038
> 0x7f8084eee0f0 – Strict RTP learning after remote address set to: 172.23.0.1:55546
<— Transmitting SIP response (1504 bytes) to WSS:10.5.152.12:41044 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;rport=41044;received=10.5.152.12;branch=z9hG4bK8666530
Call-ID: 0reku7oc329lkhvr86to
From: sip:502@my.domain.net;tag=cgttupilb1
To: sip:1003@my.domain.net;tag=VQuwunw4h9QBHaZp2qFc4-BQrTJWQOOg
CSeq: 2 INVITE
Server: Asterisk PBX 18.20.2
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, REFER, MESSAGE
Contact: sip:172.17.0.27:8089;transport=ws
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 931
v=0
o=- 3196208021 4 IN IP4 172.17.0.27
s=Asterisk
c=IN IP4 172.17.0.27
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 10058 UDP/TLS/RTP/SAVPF 9 111 0 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 F1:99:72:48:9F:3B:42:8A:07:A3:1A:08:92:01:36:B4:20:01:C1:94:99:29:00:BF:EB:24:9C:20:39:CF:89:A6
a=ice-ufrag:56e6e06272c9043d5006edca65b993dd
a=ice-pwd:11337e2e4adc359118eb59650e2f74b8
a=candidate:Hac11001b 1 UDP 2130706431 172.17.0.27 10058 typ host
a=candidate:Sc1203e80 1 UDP 1694498815 193.32.62.128 29278 typ srflx raddr 172.17.0.27 rport 10058
a=rtpmap:9 G722/8000
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
a=ssrc:982564934 cname:4dd213be-d05c-48e3-8186-1422f4c06206
a=msid:44474a52-892b-4f94-865f-5f0f84dfaa36 0986de85-f7a3-423f-8d4f-dc51d84a1f95
a=rtcp-fb:* transport-cc
a=mid:0
-- Channel PJSIP/1003-00000039 joined 'simple_bridge' basic-bridge <ae2334fe-e2ca-47ed-adc6-8f1ecacdef7b>
-- Channel PJSIP/502-00000038 joined 'simple_bridge' basic-bridge <ae2334fe-e2ca-47ed-adc6-8f1ecacdef7b>
Got RTP packet from 10.5.152.12:47670 (type 09, seq 044389, ts 1826983082, len 000160)
Sent RTP packet to 172.23.0.1:55546 (type 09, seq 031779, ts 1826983080, len 000160)
Got RTP packet from 10.5.152.12:47670 (type 09, seq 044390, ts 1826983242, len 000160)
Sent RTP packet to 172.23.0.1:55546 (type 09, seq 031780, ts 1826983240, len 000160)
<— Received SIP request (373 bytes) from WSS:10.5.152.12:41044 —>
ACK sip:172.17.0.27:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS jqtvs5k9ie78.invalid;branch=z9hG4bK810218
To: sip:1003@my.domain.net:8089;tag=VQuwunw4h9QBHaZp2qFc4-BQrTJWQOOg
From: sip:502@my.domain.net:8089;tag=cgttupilb1
CSeq: 2 ACK
Call-ID: 0reku7oc329lkhvr86to
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.20.1
Content-Length: 0
<— Received SIP request (343 bytes) from WSS:10.5.152.12:41054 —>
OPTIONS sip:501@my.domain.net:8089 SIP/2.0
Via: SIP/2.0/WSS 35kmr6o06fbn.invalid;branch=z9hG4bK6630152
To: sip:501@my.domain.net:8089
From: sip:501@my.domain.net:8089;tag=onmm8qmj4o
CSeq: 1 OPTIONS
Call-ID: 9t1g7c6i8t97d4r91j53
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.20.1
Content-Length: 0