When I make a call from webpage using WebRTC to another endpoint the call created successfully. But now one can listen. I recieve this warning
[Jun 13 01:12:11] WARNING[105672][C-00000008]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230325.18: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
<--- Received SIP request (2327 bytes) from WSS:176.43.6.98:62543 --->
INVITE sip:100@voip.erecrm.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK915249
To: <sip:100@voip.erecrm.com>
From: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
CSeq: 1 INVITE
Call-ID: i0qh9fr17kafkne255f7
Max-Forwards: 70
Contact: <sip:kr8actfe@192.0.2.202;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Type: application/sdp
Content-Length: 1746
v=0
o=mozilla...THIS_IS_SDPARTA-99.0 2084904480695185050 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 9C:7A:E8:D9:E4:05:B9:71:7A:A6:73:A2:5B:76:A7:09:BF:4D:61:17:BD:62:43:53:19:38:C0:4B:A3:46:8A:3D
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 65511 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 176.43.6.98
a=candidate:0 1 UDP 2122187007 192.168.1.2 65511 typ host
a=candidate:2 1 UDP 2122252543 172.21.176.1 65512 typ host
a=candidate:4 1 TCP 2105458943 192.168.1.2 9 typ host tcptype active
a=candidate:5 1 TCP 2105524479 172.21.176.1 9 typ host tcptype active
a=candidate:0 2 UDP 2122187006 192.168.1.2 65513 typ host
a=candidate:2 2 UDP 2122252542 172.21.176.1 65514 typ host
a=candidate:4 2 TCP 2105458942 192.168.1.2 9 typ host tcptype active
a=candidate:5 2 TCP 2105524478 172.21.176.1 9 typ host tcptype active
a=candidate:1 1 UDP 1685987327 176.43.6.98 65511 typ srflx raddr 192.168.1.2 rport 65511
a=candidate:1 2 UDP 1685987326 176.43.6.98 65513 typ srflx raddr 192.168.1.2 rport 65513
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:9993abbf2d566cdcabf8eb888e98cb15
a=ice-ufrag:21495937
a=mid:0
a=msid:{686387f3-11c7-4d66-acda-d1386d1c8f25} {881abe2c-154a-46f7-8442-c634f4fd7fb1}
a=rtcp:65513 IN IP4 176.43.6.98
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:3954319921 cname:{9cc9232e-500b-48b1-b584-cda060157aac}
<--- Transmitting SIP response (479 bytes) to WSS:176.43.6.98:62543 --->
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 192.0.2.202;rport=62543;received=176.43.6.98;branch=z9hG4bK915249
Call-ID: i0qh9fr17kafkne255f7
From: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
To: <sip:100@voip.erecrm.com>;tag=z9hG4bK915249
CSeq: 1 INVITE
WWW-Authenticate: Digest realm="voip.erecrm.com",nonce="1718230972/6e1f3f0e3570ae078b2465ca664bf249",opaque="550e373723191f25",algorithm=MD5,qop="auth"
Server: Asterisk PBX 18.21.0
Content-Length: 0
<--- Received SIP request (288 bytes) from WSS:176.43.6.98:62543 --->
ACK sip:100@voip.erecrm.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK915249
To: <sip:100@voip.erecrm.com>;tag=z9hG4bK915249
From: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
Call-ID: i0qh9fr17kafkne255f7
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0
<--- Received SIP request (2611 bytes) from WSS:176.43.6.98:62543 --->
INVITE sip:100@voip.erecrm.com SIP/2.0
Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK6684673
To: <sip:100@voip.erecrm.com>
From: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
CSeq: 2 INVITE
Call-ID: i0qh9fr17kafkne255f7
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username="111100", realm="voip.erecrm.com", nonce="1718230972/6e1f3f0e3570ae078b2465ca664bf249", uri="sip:100@voip.erecrm.com", response="98a3252c3df7a3f686dee5d898937df3", opaque="550e373723191f25", qop=auth, cnonce="bvea19nd6po8", nc=00000001
Contact: <sip:kr8actfe@192.0.2.202;transport=wss;ob>
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Type: application/sdp
Content-Length: 1746
v=0
o=mozilla...THIS_IS_SDPARTA-99.0 2084904480695185050 0 IN IP4 0.0.0.0
s=-
t=0 0
a=sendrecv
a=fingerprint:sha-256 9C:7A:E8:D9:E4:05:B9:71:7A:A6:73:A2:5B:76:A7:09:BF:4D:61:17:BD:62:43:53:19:38:C0:4B:A3:46:8A:3D
a=group:BUNDLE 0
a=ice-options:trickle
a=msid-semantic:WMS *
m=audio 65511 UDP/TLS/RTP/SAVPF 109 9 0 8 101
c=IN IP4 176.43.6.98
a=candidate:0 1 UDP 2122187007 192.168.1.2 65511 typ host
a=candidate:2 1 UDP 2122252543 172.21.176.1 65512 typ host
a=candidate:4 1 TCP 2105458943 192.168.1.2 9 typ host tcptype active
a=candidate:5 1 TCP 2105524479 172.21.176.1 9 typ host tcptype active
a=candidate:0 2 UDP 2122187006 192.168.1.2 65513 typ host
a=candidate:2 2 UDP 2122252542 172.21.176.1 65514 typ host
a=candidate:4 2 TCP 2105458942 192.168.1.2 9 typ host tcptype active
a=candidate:5 2 TCP 2105524478 172.21.176.1 9 typ host tcptype active
a=candidate:1 1 UDP 1685987327 176.43.6.98 65511 typ srflx raddr 192.168.1.2 rport 65511
a=candidate:1 2 UDP 1685987326 176.43.6.98 65513 typ srflx raddr 192.168.1.2 rport 65513
a=sendrecv
a=end-of-candidates
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:2/recvonly urn:ietf:params:rtp-hdrext:csrc-audio-level
a=extmap:3 urn:ietf:params:rtp-hdrext:sdes:mid
a=fmtp:109 maxplaybackrate=48000;stereo=1;useinbandfec=1
a=fmtp:101 0-15
a=ice-pwd:9993abbf2d566cdcabf8eb888e98cb15
a=ice-ufrag:21495937
a=mid:0
a=msid:{686387f3-11c7-4d66-acda-d1386d1c8f25} {881abe2c-154a-46f7-8442-c634f4fd7fb1}
a=rtcp:65513 IN IP4 176.43.6.98
a=rtcp-mux
a=rtpmap:109 opus/48000/2
a=rtpmap:9 G722/8000/1
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=setup:actpass
a=ssrc:3954319921 cname:{9cc9232e-500b-48b1-b584-cda060157aac}
<--- Transmitting SIP response (303 bytes) to WSS:176.43.6.98:62543 --->
SIP/2.0 100 Trying
Via: SIP/2.0/WSS 192.0.2.202;rport=62543;received=176.43.6.98;branch=z9hG4bK6684673
Call-ID: i0qh9fr17kafkne255f7
From: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
To: <sip:100@voip.erecrm.com>
CSeq: 2 INVITE
Server: Asterisk PBX 18.21.0
Content-Length: 0
-- Executing [100@internals:1] NoOp("PJSIP/111100-0000000f", "") in new stack
-- Executing [100@internals:2] Gosub("PJSIP/111100-0000000f", "subDialInternal,s,1(PJSIP/100,30)") in new stack
-- Executing [s@subDialInternal:1] Answer("PJSIP/111100-0000000f", "") in new stack
> 0x3fc5a7489000 -- Strict RTP learning after remote address set to: 176.43.6.98:65511
<--- Transmitting SIP response (1448 bytes) to WSS:176.43.6.98:62543 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.0.2.202;rport=62543;received=176.43.6.98;branch=z9hG4bK6684673
Call-ID: i0qh9fr17kafkne255f7
From: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
To: <sip:100@voip.erecrm.com>;tag=4ed5ea20-290a-11ef-b9bd-00155d016406
CSeq: 2 INVITE
Server: Asterisk PBX 18.21.0
Contact: <sip:192.168.1.170:8089;transport=ws>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 861
v=0
o=- 846524058 2 IN IP4 192.168.1.170
s=Asterisk
c=IN IP4 192.168.1.170
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE 0
m=audio 11558 UDP/TLS/RTP/SAVPF 109 9 0 101
a=connection:new
a=setup:active
a=fingerprint:SHA-256 D5:D6:9F:F3:E1:5F:B9:BB:7B:3C:CC:07:00:5A:6F:54:41:79:2D:38:BF:CF:A8:91:7D:B0:21:77:10:FF:E2:63
a=ice-ufrag:2d5cf2c76d285554576eecf060d6f56d
a=ice-pwd:0fed2b042f9cdb8b0fb0670113610bb3
a=candidate:Hb02b0662 1 UDP 2130706431 176.43.6.98 11558 typ host
a=rtpmap:109 opus/48000/2
a=fmtp:109 useinbandfec=1
a=rtpmap:9 G722/8000
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
a=ssrc:252726010 cname:d189287e-b9c9-430b-829e-e63f85564fa3
a=msid:64ec4433-ccef-4b87-821c-2b92c4fd2da6 bed16d6a-332a-46d4-ae04-f99764d19254
a=rtcp-fb:* transport-cc
a=mid:0
<--- Received SIP request (477 bytes) from WSS:176.43.6.98:62543 --->
ACK sip:192.168.1.170:8089;transport=ws SIP/2.0
Via: SIP/2.0/WSS 192.0.2.202;branch=z9hG4bK3708942
To: <sip:100@voip.erecrm.com>;tag=4ed5ea20-290a-11ef-b9bd-00155d016406
From: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
CSeq: 2 ACK
Call-ID: i0qh9fr17kafkne255f7
Max-Forwards: 70
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Length: 0
> 0x3fc5a7489000 -- Strict RTP learning after ICE completion
[Jun 13 01:22:52] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
> 0x3fc5a7489000 -- Strict RTP learning after remote address set to: 176.43.6.98:1024
[Jun 13 01:22:52] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:52] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:52] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:52] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:52] NOTICE[105718][C-0000000a]: ast_expr2.y:759 compose_func_args: argbuf allocated 5 bytes;
[Jun 13 01:22:52] NOTICE[105718][C-0000000a]: ast_expr2.y:778 compose_func_args: argbuf uses 4 bytes;
-- Executing [s@subDialInternal:2] ExecIf("PJSIP/111100-0000000f", "1?Dial(PJSIP/100,30,m):Return(BUSY)") in new stack
-- Called PJSIP/100
-- Started music on hold, class 'default', on channel 'PJSIP/111100-0000000f'
<--- Transmitting SIP request (1251 bytes) to TLS:176.43.6.98:51931 --->
INVITE sip:100@176.43.6.98:51931;transport=TLS;rinstance=e5d304b536d95bda SIP/2.0
Via: SIP/2.0/TLS 176.43.6.98:5061;rport;branch=z9hG4bKPj4f23081e-290a-11ef-b9bd-00155d016406;alias
From: "Waled Fathalla" <sip:111100@192.168.1.170>;tag=4f22e8f9-290a-11ef-b9bd-00155d016406
To: <sip:100@176.43.6.98;rinstance=e5d304b536d95bda>
Contact: <sip:asterisk@176.43.6.98:5061;transport=TLS>
Call-ID: 4f22e93a-290a-11ef-b9bd-00155d016406
CSeq: 23602 INVITE
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Supported: 100rel, timer, replaces, norefersub, histinfo
Session-Expires: 1800
Min-SE: 90
Max-Forwards: 70
User-Agent: Asterisk PBX 18.21.0
Content-Type: application/sdp
Content-Length: 481
v=0
o=- 100300561 100300561 IN IP4 176.43.6.98
s=Asterisk
c=IN IP4 176.43.6.98
t=0 0
m=audio 14070 RTP/AVP 0 8 101
a=ice-ufrag:154a9aaa4123ca0e4370a21258bdef22
a=ice-pwd:6fa28c197059d44c0e2410125f926df6
a=candidate:Hb02b0662 1 UDP 2130706431 176.43.6.98 14070 typ host
a=candidate:Hb02b0662 2 UDP 2130706430 176.43.6.98 14071 typ host
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=ptime:20
a=maxptime:140
a=sendrecv
<--- Received SIP response (359 bytes) from TLS:176.43.6.98:51931 --->
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 176.43.6.98:5061;rport=5061;branch=z9hG4bKPj4f23081e-290a-11ef-b9bd-00155d016406;alias
To: <sip:100@176.43.6.98;rinstance=e5d304b536d95bda>
From: "Waled Fathalla" <sip:111100@192.168.1.170>;tag=4f22e8f9-290a-11ef-b9bd-00155d016406
Call-ID: 4f22e93a-290a-11ef-b9bd-00155d016406
CSeq: 23602 INVITE
Content-Length: 0
<--- Received SIP response (698 bytes) from TLS:176.43.6.98:51931 --->
SIP/2.0 180 Ringing
Via: SIP/2.0/TLS 176.43.6.98:5061;rport=5061;branch=z9hG4bKPj4f23081e-290a-11ef-b9bd-00155d016406;alias
Contact: <sip:100@176.43.6.98:51931;transport=TLS>
To: <sip:100@176.43.6.98;rinstance=e5d304b536d95bda>;tag=6d879007
From: "Waled Fathalla" <sip:111100@192.168.1.170>;tag=4f22e8f9-290a-11ef-b9bd-00155d016406
Call-ID: 4f22e93a-290a-11ef-b9bd-00155d016406
CSeq: 23602 INVITE
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 0
-- PJSIP/100-00000010 is ringing
[Jun 13 01:22:52] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:52] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:53] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:53] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:54] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:54] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
<--- Received SIP response (972 bytes) from TLS:176.43.6.98:51931 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 176.43.6.98:5061;rport=5061;branch=z9hG4bKPj4f23081e-290a-11ef-b9bd-00155d016406;alias
Require: timer
Contact: <sip:100@176.43.6.98:51931;transport=TLS>
To: <sip:100@176.43.6.98;rinstance=e5d304b536d95bda>;tag=6d879007
From: "Waled Fathalla" <sip:111100@192.168.1.170>;tag=4f22e8f9-290a-11ef-b9bd-00155d016406
Call-ID: 4f22e93a-290a-11ef-b9bd-00155d016406
CSeq: 23602 INVITE
Session-Expires: 1800;refresher=uac
Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE
Content-Type: application/sdp
Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri
User-Agent: Zoiper v2.10.20.4_1
Allow-Events: presence, kpml, talk, as-feature-event
Content-Length: 193
v=0
o=Zoiper 0 56933244 IN IP4 176.43.6.98
s=Zoiper
c=IN IP4 176.43.6.98
t=0 0
m=audio 38439 RTP/AVP 0 8 3 101
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=sendrecv
a=rtcp-mux
> 0x3fc5a74c8000 -- Strict RTP learning after remote address set to: 176.43.6.98:38439
<--- Transmitting SIP request (450 bytes) to TLS:176.43.6.98:51931 --->
ACK sip:100@176.43.6.98:51931;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 176.43.6.98:5061;rport;branch=z9hG4bKPj507caef6-290a-11ef-b9bd-00155d016406;alias
From: "Waled Fathalla" <sip:111100@192.168.1.170>;tag=4f22e8f9-290a-11ef-b9bd-00155d016406
To: <sip:100@176.43.6.98;rinstance=e5d304b536d95bda>;tag=6d879007
Call-ID: 4f22e93a-290a-11ef-b9bd-00155d016406
CSeq: 23602 ACK
Max-Forwards: 70
User-Agent: Asterisk PBX 18.21.0
Content-Length: 0
-- PJSIP/100-00000010 answered PJSIP/111100-0000000f
-- Stopped music on hold on PJSIP/111100-0000000f
-- Channel PJSIP/100-00000010 joined 'simple_bridge' basic-bridge <249e73d1-f28f-4c43-ade8-e029c811b40b>
> 0x3fc5a74c8000 -- Strict RTP switching to RTP target address 176.43.6.98:38439 as source
-- Channel PJSIP/111100-0000000f joined 'simple_bridge' basic-bridge <249e73d1-f28f-4c43-ade8-e029c811b40b>
[Jun 13 01:22:58] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
[Jun 13 01:22:58] WARNING[105718][C-0000000a]: res_rtp_asterisk.c:3255 __rtp_recvfrom: ElmasCallCenter-1718230972.24: DTLS packet from 176.43.6.98:1024 dropped. Source not in ICE active candidate list.
> 0x3fc5a74c8000 -- Strict RTP learning complete - Locking on source address 176.43.6.98:38439
<--- Received SIP request (485 bytes) from TLS:176.43.6.98:51931 --->
BYE sip:asterisk@176.43.6.98:5061;transport=TLS SIP/2.0
Via: SIP/2.0/TLS 192.168.1.9:60455;branch=z9hG4bK-524287-1---522fb9cb5ab6e003;rport
Max-Forwards: 70
Contact: <sip:100@176.43.6.98:51931;transport=TLS>
To: "Waled Fathalla" <sip:111100@192.168.1.170>;tag=4f22e8f9-290a-11ef-b9bd-00155d016406
From: <sip:100@176.43.6.98;rinstance=e5d304b536d95bda>;tag=6d879007
Call-ID: 4f22e93a-290a-11ef-b9bd-00155d016406
CSeq: 2 BYE
User-Agent: Zoiper v2.10.20.4_1
Content-Length: 0
<--- Transmitting SIP response (399 bytes) to TLS:176.43.6.98:51931 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 192.168.1.9:60455;rport=51931;received=176.43.6.98;branch=z9hG4bK-524287-1---522fb9cb5ab6e003
Call-ID: 4f22e93a-290a-11ef-b9bd-00155d016406
From: <sip:100@176.43.6.98;rinstance=e5d304b536d95bda>;tag=6d879007
To: "Waled Fathalla" <sip:111100@192.168.1.170>;tag=4f22e8f9-290a-11ef-b9bd-00155d016406
CSeq: 2 BYE
Server: Asterisk PBX 18.21.0
Content-Length: 0
-- Channel PJSIP/100-00000010 left 'simple_bridge' basic-bridge <249e73d1-f28f-4c43-ade8-e029c811b40b>
-- Channel PJSIP/111100-0000000f left 'simple_bridge' basic-bridge <249e73d1-f28f-4c43-ade8-e029c811b40b>
== Spawn extension (subDialInternal, s, 2) exited non-zero on 'PJSIP/111100-0000000f'
<--- Transmitting SIP request (448 bytes) to WSS:176.43.6.98:62543 --->
BYE sip:kr8actfe@176.43.6.98:62543;transport=ws;ob SIP/2.0
Via: SIP/2.0/WSS 192.168.1.170:8089;rport;branch=z9hG4bKPj53f93ef2-290a-11ef-b9bd-00155d016406;alias
From: <sip:100@voip.erecrm.com>;tag=4ed5ea20-290a-11ef-b9bd-00155d016406
To: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
Call-ID: i0qh9fr17kafkne255f7
CSeq: 27873 BYE
Reason: Q.850;cause=16
Max-Forwards: 70
User-Agent: Asterisk PBX 18.21.0
Content-Length: 0
<--- Received SIP response (480 bytes) from WSS:176.43.6.98:62543 --->
SIP/2.0 200 OK
Via: SIP/2.0/WSS 192.168.1.170:8089;rport;branch=z9hG4bKPj53f93ef2-290a-11ef-b9bd-00155d016406;alias
From: <sip:100@voip.erecrm.com>;tag=4ed5ea20-290a-11ef-b9bd-00155d016406
To: "Waled Fathalla" <sip:111100@voip.erecrm.com>;tag=kk2thdfpiv
CSeq: 27873 BYE
Call-ID: i0qh9fr17kafkne255f7
Supported: outbound
User-Agent: Browser Phone 0.3.27 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64; rv:126.0) Gecko/20100101 Firefox/126.0
Content-Length: 0
The message appears to be correct. There is a problem with the sender, or possibly with a router. According to the SDP, the port should be 65511 or 65513, but was actually 1024.
I couldn’t solve the problem yet. Is there anyone who advises me what to do?
Examine the actual ICE negotiation and the flow of the traffic, and see if it truly is coming from a source that is not in any of the ICE candidates. If that’s the case there’s not really anything you could do, except modify the code I guess to remove the denial of service protection for that scenario.
The router configuration is just port forward to ports 5060 and 5061 and 10000-20000. I understand you reply. do you have any advice to resolve this problem?
Thanks alot
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