Call disconnection after 32 seconds in whatsapp sip calling integration with asterisk

Asterisk Asterisk SIP
1

Asterisk 22.5 is sending 200 OK responses over TLS to incoming SIP calls, but the remote side never sends an ACK. The call seems to connect briefly for a few seconds, then Asterisk retries several times, eventually sending a BYE due to timeout, and receives a 404 Not Found in response.

<— Transmitting SIP response (200 OK) to TLS:remote.server.example:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS [remote IP]:5061;rport=5061;branch=…
From: sip:+1234567890@wp.meta.vc;tag=…
To: sip:+1987654321@mydomain.example;tag=…
CSeq: 100 INVITE
Server: Asterisk PBX 22.5.0
Contact: sip:+1987654321@203.0.113.10:5061;transport=TLS
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759496847581 4 IN IP4 203.0.113.10
s=Asterisk
c=IN IP4 203.0.113.10
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 11874 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 17:22:5C:08:97:19:F0:2F:E8:AD:BF:62:44:6B:57:2C:15:C4:79:FA:41:27:32:KT:13:40:14:BA:7F:F5:C5:88
a=ice-ufrag:5354f1b5799b8d8652840e8a0c5db680
a=ice-pwd:41e3856e125f5cc865279af92f36b663
a=candidate:H14c41a9a 1 UDP 2130706431 203.0.113.10 11874 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 11874 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 11874 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:2132923805 cname:fa73a679-7b03-4693-842b-f3dcabc212d6
a=msid:c948455c-ec0e-4613-9cf1-19cd7f00a0af 51741300-26ad-439b-bb9d-f21b7762b811
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

Asterisk retires above response several times then sends bye message

<— Transmitting SIP request (BYE) to TLS:remote.server.example:5061 —>
BYE sip:+1234567890@remote.domain;transport=tls SIP/2.0
Via: SIP/2.0/TLS [203.0.113.10]:5061;rport;branch=…
From: sip:+1987654321@mydomain.example;tag=…
To: sip:+1234567890@remote.domain;tag=…
Call-ID: …
CSeq: 101 BYE
Reason: SIP; cause=408; text=“Request Timeout”

<— Received SIP response —>
SIP/2.0 404 Not Found

Also i can see this error in cli

ssl0x7abed80162a0 Error reading CA certificates from buffer

2

The log is incomplete, and contains an unusual port number for a TCP source address.

3

<— Transmitting SIP response (2235 bytes) to TLS:69.171.251.114:41010 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=41010;received=69.171.251.114;branch=z9hG4bKPj89a9310503bdde17a8a5132779e1ef0f;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj89a9310503bdde17a8a5132779e1ef0f
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:63445;rport=63445;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjJtMyoe9E5XXYdZZXEl782e4-BFEXjKzy;alias
Record-Route: sip:69.171.251.114:41010;transport=TLS;lr
Record-Route: sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNFRkM5NUFBOUI5NzcwMDNGMTlDQTBEMkEwMzY5QkEcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “919876543210” sip:+919876543210@wa.meta.vc;tag=e6954639-8196-4fab-b258-e1a9dc602942
To: sip:+971500000000@newsip.example.com;tag=271c9668-243f-4c35-923e-128188d9a093
CSeq: 30609 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: sip:971500000000@45.123.67.89:5061;transport=TLS
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759496847581 4 IN IP4 45.123.67.89
s=Asterisk
c=IN IP4 45.123.67.89
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 11874 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 17:22:5C:08:56:19:F0:2F:E8:AD:AB:62:44:6B:57:2C:15:C4:34:FA:41:27:32:BC:13:40:14:BA:7F:F5:C5:88
a=ice-ufrag:5354f1b5799b8d8652840e8a0c5db680
a=ice-pwd:41e3856e125f5cc865279af92f36b663
a=candidate:H14c41a9a 1 UDP 2130706431 45.123.67.89 11874 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 11874 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 11874 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:2132923805 cname:fa73a679-7b03-4693-842b-f3yutti212d6
a=msid:c948455c-ec0e-9778-9cf1-19cd7f00a0af 51741300-26ad-439b-bb9d-f21b7762b811
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

<— Transmitting SIP request (720 bytes) to TLS:69.171.251.114:41010 —>
BYE sip:+919876543210@wa.meta.vc;transport=tls;ob SIP/2.0
Via: SIP/2.0/TLS 45.123.67.89:5061;rport;branch=z9hG4bKPj0c4e1938-a272-483f-b2c3-cc65519ea8af;alias
From: sip:+971500000000@newsip.example.com;tag=271c9668-243f-4c35-923e-128188d9a093
To: “919876543210” sip:+919876543210@wa.meta.vc;tag=e6954639-8196-4fab-b258-e1a9dc602942
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNFRkM5NUFBOUI5NzcwMDNGMTlDQTBEMkEwMzY5QkEcGAw5NzE1ODUxMDM0NzIVAgAVGgA
CSeq: 19865 BYE
Route: sip:69.171.251.114:41010;transport=TLS;lr
Route: sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr
Reason: SIP ;cause=408 ;text=“Request Timeout”
Max-Forwards: 70
User-Agent: Asterisk PBX 22.5.0
Content-Length: 0

<— Received SIP response (497 bytes) from TLS:69.171.251.114:41010 —>
SIP/2.0 404 Not Found
Via: SIP/2.0/TLS 45.123.67.89:5061;rport=5061;received=45.123.67.89;branch=z9hG4bKPj0c4e1938-a272-483f-b2c3-cc65519ea8af;alias
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNFRkM5NUFBOUI5NzcwMDNGMTlDQTBEMkEwMzY5QkEcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: sip:+971500000000@newsip.example.com;tag=271c9668-243f-4c35-923e-128188d9a093
To: “919876543210” sip:+919876543210@wa.meta.vc;tag=e6954639-8196-4fab-b258-e1a9dc602942
CSeq: 19865 BYE
Content-Length: 0

These are the actual Asterisk CLI logs captured during the SIP call flow.
I have removed all sensitive information for privacy purposes

4

A SIP call starts with an INVITE.

Also please use /var/log/asterisk/full, not a screen scrape, and use </> to mark the log as preformatted text, for the forum.

5

i also faced this issue already . When Whatsapp don’t accept RTP from asterisk server then it will either hear no-audio or call disconnect after 30 sec. so i suggest you to check tls domain name and pointing server. domain/sub-domain you are using to make tls connection should point to asterisk server only otherwise whatsapp will no allow or accept any RTP. and main thing is whatsapp do not allow wildcard certificate so use same domain/sub-domain in CN name as well.
like here you are using mydomain.example.com
then CN name should CN= mydomain.example.com
don’t use CN= *.example.com

6

[Oct 6 09:51:05] VERBOSE[37] res_pjsip_logger.c: <— Received SIP request (2154 bytes) from TLS:69.171.251.115:51134 —>
INVITE sip:+971897451254@developer.sip.com:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 31.13.66.215:5061;rport;branch=z9hG4bKPjeb00299eeb617a8502b749a9005e07f5;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPjeb00299eeb617a8502b749a9005e07f5
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPj4IxALRZwfKoEYVXu4-nkph9OAye7oHki;alias
Max-Forwards: 69
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>
Contact: <sip:+917859745878@wa.meta.vc;transport=tls;ob>;isfocus
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
CSeq: 32371 INVITE
X-FB-External-Domain: wa.meta.vc
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
User-Agent: Facebook SipGateway
Content-Type: application/sdp
Content-Length: 1030

v=0
o=- 1759744264890 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 05e0d691-acd3-4c01-a543-8ea80b321c3d
a=ice-lite
m=audio 3484 UDP/TLS/RTP/SAVPF 111 126
c=IN IP4 157.240.239.52
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2985973821 1 udp 2122260223 157.240.239.52 3484 typ host generation 0 network-cost 50
a=candidate:429436147 1 udp 2122262783 2a03:2880:f244:1c5:face:b00c:0:699c 3484 typ host generation 0 network-cost 50
a=ice-ufrag:QcwmphUvDAdDRBIk
a=ice-pwd:RWKMn2uMaWEV2SHsitfnEQ==
a=fingerprint:sha-256 AC:B3:99:68:36:0B:F3:84:5D:C5:88:0B:72:9F:4B:87:64:C2:F5:17:16:CA:B3:3D:87:B3:97:0C:04:24:5C:96
a=setup:actpass
a=mid:audio
a=sendrecv
a=msid:05e0d691-acd3-4c01-a543-8ea80b321c3d WhatsAppTrack1
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=maxptime:20
a=ptime:20
a=ssrc:1200038226 cname:WhatsAppAudioStream1

[Oct 6 09:51:07] VERBOSE[38] res_pjsip_logger.c: <— Transmitting SIP response (1093 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPjeb00299eeb617a8502b749a9005e07f5;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPjeb00299eeb617a8502b749a9005e07f5
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPj4IxALRZwfKoEYVXu4-nkph9OAye7oHki;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=z9hG4bKPjeb00299eeb617a8502b749a9005e07f5
CSeq: 32371 INVITE
WWW-Authenticate: Digest realm=“asterisk”,nonce=“1759744267/7a070950ee7bf5f5ba4c953f6ccbdfdc”,opaque=“76308be45066b3f0”,algorithm=MD5,qop=“auth”
Server: Asterisk PBX 22.5.0
Content-Length: 0

[Oct 6 09:51:07] VERBOSE[37] res_pjsip_logger.c: <— Received SIP request (534 bytes) from TLS:69.171.251.115:51134 —>
ACK sip:+971897451254@developer.sip.com:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 31.13.66.215:5061;rport;branch=z9hG4bKPjeb00299eeb617a8502b749a9005e07f5;alias
Max-Forwards: 70
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=z9hG4bKPjeb00299eeb617a8502b749a9005e07f5
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
CSeq: 32371 ACK
Content-Length: 0

[Oct 6 09:51:07] VERBOSE[37] res_pjsip_logger.c: <— Received SIP request (2474 bytes) from TLS:69.171.251.115:51134 —>
INVITE sip:+971897451254@developer.sip.com:5061;transport=tls SIP/2.0
Via: SIP/2.0/TLS 31.13.66.215:5061;rport;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Max-Forwards: 69
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>
Contact: <sip:+917859745878@wa.meta.vc;transport=tls;ob>;isfocus
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
CSeq: 32372 INVITE
X-FB-External-Domain: wa.meta.vc
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
User-Agent: Facebook SipGateway
Authorization: Digest username=“971897451254”, realm=“asterisk”, nonce=“1759744267/7a070950ee7bf5f5ba4c953f6ccbdfdc”, uri=“sip:+971897451254@developer.sip.com:5061”, response=“00b35305c1e01c18d4e6fbe7615c30c6”, algorithm=MD5, cnonce=“VT5vbSAG4h5pEGjUFhVAiGlJGLKms7v”, opaque=“76308be45066b3f0”, qop=auth, nc=00000001
Content-Type: application/sdp
Content-Length: 1030

v=0
o=- 1759744264890 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 05e0d691-acd3-4c01-a543-8ea80b321c3d
a=ice-lite
m=audio 3484 UDP/TLS/RTP/SAVPF 111 126
c=IN IP4 157.240.239.52
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2985973821 1 udp 2122260223 157.240.239.52 3484 typ host generation 0 network-cost 50
a=candidate:429436147 1 udp 2122262783 2a03:2880:f244:1c5:face:b00c:0:699c 3484 typ host generation 0 network-cost 50
a=ice-ufrag:QcwmphUvDAdDRBIk
a=ice-pwd:RWKMn2uMaWEV2SHsitfnEQ==
a=fingerprint:sha-256 AC:B3:99:68:36:0B:F3:84:5D:C5:88:0B:72:9F:4B:87:64:C2:F5:17:16:CA:B3:3D:87:B3:97:0C:04:24:5C:96
a=setup:actpass
a=mid:audio
a=sendrecv
a=msid:05e0d691-acd3-4c01-a543-8ea80b321c3d WhatsAppTrack1
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=rtcp-fb:111 transport-cc
a=fmtp:111 maxaveragebitrate=20000;maxplaybackrate=16000;minptime=20;sprop-maxcapturerate=16000;useinbandfec=1
a=rtpmap:126 telephone-event/8000
a=maxptime:20
a=ptime:20
a=ssrc:1200038226 cname:WhatsAppAudioStream1

[Oct 6 09:51:12] VERBOSE[38] res_pjsip_logger.c: <— Transmitting SIP response (895 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 100 Trying
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Content-Length: 0

[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:1] NoOp(“PJSIP/971897451254_897451254-0000000b”, “Incoming call on +971897451254”)
[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:2] Progress(“PJSIP/971897451254_897451254-0000000b”, “”)
[Oct 6 09:51:12] VERBOSE[38] res_rtp_asterisk.c: 0x7b6b0814b3f0 – Strict RTP learning after remote address set to: 157.240.239.52:3484
[Oct 6 09:51:12] VERBOSE[38] res_pjsip_logger.c: <— Transmitting SIP response (2186 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Contact: <sip:971897451254@30.125.26.145;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:3] Set(“PJSIP/971897451254_897451254-0000000b”, “DIVERSION=”)
[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:4] Set(“PJSIP/971897451254_897451254-0000000b”, “TEMP1=”)
[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:5] Set(“PJSIP/971897451254_897451254-0000000b”, “TEMP2=”)
[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:6] Set(“PJSIP/971897451254_897451254-0000000b”, “TEMP2=”)
[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:7] Set(“PJSIP/971897451254_897451254-0000000b”, “CALLED_NUMBER=+971897451254”)
[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:8] NoOp(“PJSIP/971897451254_897451254-0000000b”, “Called Number: +971897451254”)
[Oct 6 09:51:12] VERBOSE[653][C-00000007] pbx_realtime.c: Executing [+971897451254@from-twilio:9] Stasis(“PJSIP/971897451254_897451254-0000000b”, “webrtc,called_number=+971897451254”)

[Oct 6 09:51:13] VERBOSE[38] res_pjsip_logger.c: <— Transmitting SIP request (450 bytes) to TLS:163.70.144.192:5061 —>
OPTIONS sip:wa.meta.vc:5061 SIP/2.0
Via: SIP/2.0/TLS 30.125.26.145:5061;rport;branch=z9hG4bKPj48e53357-55d6-4e1c-a521-eb5cd829ab9b;alias
From: <sip:971897451254@developer.sip.com>;tag=77afe4d0-2ead-4638-924d-da83db0d7274
To: <sip:wa.meta.vc>
Contact: <sip:971897451254@30.125.26.145:5061;transport=TLS>
Call-ID: 93c5eb8d-be9c-41d1-a911-d7d6efdff0dc
CSeq: 4009 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 22.5.0
Content-Length: 0

[Oct 6 09:51:13] VERBOSE[653][C-00000007] res_rtp_asterisk.c: 0x7b6b0814b3f0 – Strict RTP learning after ICE completion
[Oct 6 09:51:13] VERBOSE[50] res_rtp_asterisk.c: 0x7b6b0814b3f0 – Strict RTP learning after remote address set to: 157.240.239.52:3484

[Oct 6 09:51:13] ERROR[37] pjproject: tlsc0x7b6b08268528 RFC 5922 (section 7.2) does not allow TLS wildcard certificates. Advise your SIP provider, please!
[Oct 6 09:51:13] NOTICE[37] res_pjsip/pjsip_transport_events.c: Transport ‘transport-tls’ to remote ‘wa.meta.vc’ - 163.70.144.192:5061 - The certificate is untrusted
[Oct 6 09:51:14] VERBOSE[37] res_pjsip_logger.c: <— Received SIP response (567 bytes) from TLS:163.70.144.192:5061 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 30.125.26.145:5061;rport=55484;received=2803:6080:9120:8b90:9c52:4d6f:400:0;branch=z9hG4bKPj48e53357-55d6-4e1c-a521-eb5cd829ab9b;alias
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Call-ID: 93c5eb8d-be9c-41d1-a911-d7d6efdff0dc
From: <sip:971897451254@developer.sip.com>;tag=77afe4d0-2ead-4638-924d-da83db0d7274
To: <sip:wa.meta.vc>;tag=z9hG4bKPjfa9f5d8d43aee61d7a1a301ee17f01c7
CSeq: 4009 OPTIONS
X-FB-External-Domain: wa.meta.vc
Content-Length: 0

[Oct 6 09:51:19] VERBOSE[195] res_rtp_asterisk.c: 0x7b6ad0090b10 – Strict RTP qualifying stream type:
[Oct 6 09:51:19] VERBOSE[195] res_rtp_asterisk.c: 0x7b6ad0090b10 – Strict RTP qualifying stream type:
[Oct 6 09:51:19] VERBOSE[195] res_rtp_asterisk.c: 0x7b6ad0090b10 – Strict RTP qualifying stream type:
[Oct 6 09:51:19] VERBOSE[195] res_rtp_asterisk.c: 0x7b6ad0090b10 – Strict RTP qualifying stream type:
[Oct 6 09:51:19] VERBOSE[195] res_rtp_asterisk.c: 0x7b6ad0090b10 – Strict RTP switching source address to 10.0.0.66:54774
[Oct 6 09:51:19] VERBOSE[195] res_rtp_asterisk.c: 0x7b6ad0090b10 – Strict RTP learning complete - Locking on source address 10.0.0.66:54774
[Oct 6 09:51:19] VERBOSE[38] res_pjsip_logger.c: <— Transmitting SIP response (2220 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971897451254@30.125.26.145;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

[Oct 6 09:51:19] NOTICE[38] res_pjsip/config_transport.c: Transport ‘transport-tcp-nat’ is not fully reloadable, not reloading: protocol, bind, TLS (everything but certificate and private key if filename is unchanged), TCP, ToS, or CoS options.
[Oct 6 09:51:19] NOTICE[38] res_pjsip/config_transport.c: Transport ‘transport-tcp-nat’ is not fully reloadable, not reloading: protocol, bind, TLS (everything but certificate and private key if filename is unchanged), TCP, ToS, or CoS options.
[Oct 6 09:51:19] VERBOSE[38] res_pjsip_logger.c: <— Transmitting SIP request (511 bytes) to TCP:54.172.60.3:5060 —>
OPTIONS sip:testtwiliopronnel.pstn.twilio.com:5060 SIP/2.0
Via: SIP/2.0/TCP 30.125.26.145:5060;rport;branch=z9hG4bKPj72848d6c-8a97-438d-9bf4-43d3ddf28a0d;alias
From: <sip:+18647148595@testtwiliopronnel.pstn.twilio.com>;tag=b1686651-bbe7-460e-a186-7cae45dc29b9
To: <sip:testtwiliopronnel.pstn.twilio.com>
Contact: <sip:+18647148595@30.125.26.145:5060;transport=TCP>
Call-ID: 85e6fae6-7d3b-4121-b777-e38221723ec3
CSeq: 62182 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 22.5.0
Content-Length: 0

[Oct 6 09:51:20] VERBOSE[656][C-00000007] bridge_channel.c: Channel PJSIP/971897451254_897451254-0000000b joined ‘simple_bridge’ stasis-bridge <71952370-ef69-4412-bf3f-95bff0a09268>
[Oct 6 09:51:20] VERBOSE[37] res_pjsip_logger.c: <— Received SIP response (452 bytes) from TCP:54.172.60.3:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/TCP 30.125.26.145:5060;rport=57577;branch=z9hG4bKPj72848d6c-8a97-438d-9bf4-43d3ddf28a0d;alias;received=30.125.26.145
From: <sip:+18647148595@testtwiliopronnel.pstn.twilio.com>;tag=b1686651-bbe7-460e-a186-7cae45dc29b9
To: <sip:testtwiliopronnel.pstn.twilio.com>;tag=31ea4de17459eae6756773322c547ef7.f59e564d
Call-ID: 85e6fae6-7d3b-4121-b777-e38221723ec3
CSeq: 62182 OPTIONS
Server: Twilio Gateway
Content-Length: 0

[Oct 6 09:51:20] VERBOSE[657] bridge_channel.c: Channel UnicastRTP/10.0.0.66:11590-0x7b6ad0047ae0 joined ‘simple_bridge’ stasis-bridge <71952370-ef69-4412-bf3f-95bff0a09268>
[Oct 6 09:51:20] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP response (2220 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971897451254@30.125.26.145;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

7

Oct 6 09:51:20] NOTICE[37] res_pjsip/config_transport.c: Transport ‘transport-udp-nat’ is not fully reloadable, not reloading: protocol, bind, TLS (everything but certificate and private key if filename is unchanged), TCP, ToS, or CoS options.
[Oct 6 09:51:20] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP request (410 bytes) to UDP:142.251.220.46:5060 —>
OPTIONS sip:google.com:5060 SIP/2.0
Via: SIP/2.0/UDP 30.125.26.145:5060;rport;branch=z9hG4bKPjd5cd8898-581c-4e90-be14-56f2fc57b4d2
From: <sip:tarunn@google.com>;tag=af32ffa1-9947-4ffd-886a-f201556aff88
To: <sip:google.com>
Contact: <sip:tarunn@30.125.26.145:5060>
Call-ID: e392f0bc-ee21-4f17-9030-168185635a8b
CSeq: 14961 OPTIONS
Max-Forwards: 70
User-Agent: Asterisk PBX 22.5.0
Content-Length: 0

[Oct 6 09:51:21] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP response (2216 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971897451254@10.2.0.11``;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

[Oct 6 09:51:23] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP response (2216 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971897451254@10.2.0.11``;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

[Oct 6 09:51:25] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP response (2177 bytes) to TLS:69.171.251.7:65076 —>
SIP/2.0 183 Session Progress
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=65076;received=69.171.251.7;branch=z9hG4bKPj4a9624804fca0f65ce1e1eb1dbe3e9bb;alias
Via: SIP/2.0/TLS [2803:6080:e870:eb3:1b82:456e:1485:a00]:5061;branch=z9hG4bKPj4a9624804fca0f65ce1e1eb1dbe3e9bb
Via: SIP/2.0/TLS [2803:6080:e870:eb3:4e22:2c6b:1466:a00]:49587;rport=49587;received=2803:6080:e870:eb3:4e22:2c6b:1466:a00;branch=z9hG4bKPjqoKtKINASvPKccQBud3N32wdDrGJx0DW;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNENTZGQUFBQzEwMkM1RDk3NkM2MEM0RjlDOEVFQTQcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=0c8757e5-ec35-4ff0-bf9d-5b665e53f0e4
To: <sip:+971897451254@developer.sip.com>;tag=862b467d-7461-4b62-bcea-502a407f1a8c
CSeq: 13724 INVITE
Server: Asterisk PBX 22.5.0
Contact: <sip:971897451254@10.2.0.11``;transport=TLS>
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759740915990 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 12320 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 01:2C:C1:9A:09:DC:9C:A2:20:FC:50:8A:61:27:D6:F3:C4:53:44:57:52:36:13:AA:0F:58:91:BC:65:D3:A0:70
a=ice-ufrag:5d3a1df6493696112f4a26d058a20955
a=ice-pwd:6566597a55d9e4b246312c8c57680aca
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 12320 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 12320 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 12320 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1989688013 cname:e1f73746-9692-48da-b417-003e5bc428f9
a=msid:c8c6953c-ac64-4b59-9257-2f7305718370 72b1f729-3f38-4639-899e-57133d9c26f9
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

[Oct 6 09:51:27] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP response (2216 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971897451254@10.2.0.11``;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

[Oct 6 09:51:29] VERBOSE[656][C-00000007] res_rtp_asterisk.c: 0x7b6b0814b3f0 – Strict RTP switching to RTP target address 157.240.239.52:3484 as source
[Oct 6 09:51:29] VERBOSE[656][C-00000007] res_rtp_asterisk.c: 0x7b6b0814b3f0 – Strict RTP learning complete - Locking on source address 157.240.239.52:3484

[Oct 6 09:51:31] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP response (2216 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971897451254@10.2.0.11``;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

[Oct 6 09:51:35] VERBOSE[37] res_pjsip_logger.c: <— Transmitting SIP response (2216 bytes) to TLS:69.171.251.115:51134 —>
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=51134;received=69.171.251.115;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj0e8673e36313f71c0822886113333e96
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPjOtTIs0r0-qw3EBabKT8o4Fhxst8d6nyR;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: “917859745878” <sip:+917859745878@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971897451254@developer.sip.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32372 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971897451254@10.2.0.11``;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length: 1032

v=0
o=- 1759744264890 4 IN IP4 30.125.26.145
s=Asterisk
c=IN IP4 30.125.26.145
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 18562 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 9C:38:AE:98:FF:9F:18:15:EA:B2:2F:02:09:F4:56:A0:B3:68:DE:A4:8A:32:AA:C4:74:1F:6D:B8:5B:0A:CE:60
a=ice-ufrag:732fa5bc511136fe3305914e1dd0c3b8
a=ice-pwd:23b0a4f766fdf27e1ed0c9e23f677519
a=candidate:H14c41a9a 1 UDP 2130706431 30.125.26.145 18562 typ host
a=candidate:Hac110001 1 UDP 2130706431 172.17.0.1 18562 typ host
a=candidate:Ha0000da 1 UDP 2130706431 10.0.0.218 18562 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:1374662252 cname:161fbe78-031f-4759-a908-e5fe1b0d553b
a=msid:82c833ac-25a0-4bf2-8cfa-d396bc890d8e 77a3be05-08d8-464a-9582-bca941d3d0a3
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

9

I have shared the required logs

Bye message and reply is below

[Oct  6 09:51:59] VERBOSE[37] res_pjsip_logger.c: <--- Received SIP request (1047 bytes) from TLS:69.171.251.7:38576 --->
BYE sip:971585103472@20.196.26.154;transport=tls SIP/2.0
Via: SIP/2.0/TLS 31.13.66.215:5061;rport;branch=z9hG4bKPj535dd03897e726a213f6f8edee1294c8;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj535dd03897e726a213f6f8edee1294c8
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPj6hqGX4NNxJr3AGCwmjsb3G-ATtRDGcv0;alias
Max-Forwards: 69
From: "918957141005" <sip:+918957141005@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971585103472@dev2sip.pronnel.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
CSeq: 32373 BYE
X-FB-External-Domain: wa.meta.vc
Allow: INVITE, ACK, BYE, CANCEL, NOTIFY, OPTIONS
User-Agent: Facebook SipGateway
Content-Length:  0


[Oct  6 09:51:59] VERBOSE[37] res_pjsip_logger.c: <--- Transmitting SIP response (956 bytes) to TLS:69.171.251.7:38576 --->
SIP/2.0 481 Call/Transaction Does Not Exist
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=38576;received=69.171.251.7;branch=z9hG4bKPj535dd03897e726a213f6f8edee1294c8;alias
Via: SIP/2.0/TLS [2803:6080:f948:9597:b373:3c65:148e:a00]:5061;branch=z9hG4bKPj535dd03897e726a213f6f8edee1294c8
Via: SIP/2.0/TLS [2803:6080:f97c:2ab2:821f:4064:1453:a00]:46061;rport=46061;received=2803:6080:f97c:2ab2:821f:4064:1453:a00;branch=z9hG4bKPj6hqGX4NNxJr3AGCwmjsb3G-ATtRDGcv0;alias
Record-Route: <sip:wa.meta.vc;transport=tls;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUNDNzI5REFFNURGMjNENjRFQzQxRDRGNUY1RjNFNDMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: "918957141005" <sip:+918957141005@wa.meta.vc>;tag=bdf20632-0d31-42c1-92fe-838b6d166507
To: <sip:+971585103472@dev2sip.pronnel.com>;tag=6b784d9b-f196-4725-b3c7-1f7ba9a2386d
CSeq: 32373 BYE
Server: Asterisk PBX 22.5.0
Content-Length:  0


type or paste code here
10

CN for my domain is exactly same as i am using

11 
[transport-tls]

type=transport

protocol=tls

bind=0.0.0.0:5061

external_signaling_address=<SERVER_IP>

external_media_address=<SERVER_IP>

cert_file=/etc/asterisk/keys/mydomain.com/mydomain.crt

priv_key_file=/etc/asterisk/keys/mydomain.com/mydomain.key

ca_list_file=/etc/asterisk/keys/mydomain.com/mydomain.ca-bundle

method=tlsv1_2

verify_client=no

require_client_cert=no

allow_reload=yes

cipher=ECDHE-RSA-AES256-GCM-SHA384



[meta-endpoint-in]

type=endpoint

transport=transport-tls

context=from-meta

disallow=all

allow=opus

aors=meta-aor-in

auth=meta-auth-in             

rtcp_mux=yes

media_encryption=sdes



[meta-aor-in]

type=aor

max_contacts=1



[meta-auth-in]

type=auth

auth_type=userpass

username=91whatsappnumber

password=lXXXXXXXXXXXXXXj2 ------given by WABA ID



[meta-identify-in]

type=identify

endpoint=meta-endpoint-in

match=0.0.0.0/0

refer this replaced your parameter
and check domain certificate by using openssl_client command

12

You are transmitting a local address in the Contact header, which generally means you’ve failed to configure a public signalling address. Oddly you do have a public media address. Could there be a typo in your signalling address line?

(The documentation is confusing as some of the textual descriptions use the British spelling, of signalling, with two l’s, but the same document uses the American, single l, spelling in the parameter name.)

The port numbers here look more sensible.

This the wrong way round. The SIP specification explicitly bars wildcards, and Asterisk doesn’t accept them, but Meta send them anyway, so you have to disable server identity checking, which is what is happening, anyway.

13

Actually you haven’t specified local_net, although I don’t understand why that is only affecting the signalling address.

14

Call is working perfectly but only till 32 seconds, as ACK message never received
Audio is working properly two way until bye message

i have specified local_net in my transport but no effect

15

Until you get it to send the correct Contact header value (the public address) you are not going to receive that ACK, as the ACK will be sent to the address in the Contact header. As such you can forget the lack of ACK and concentrate on the symptom closer to the root cause.

One thing I would suggest is using the CLI to query the actual configuration.

Did you do a complete restart after adding local_net?

I suppose there could be a problem of handling IPV6 in a NAT environment.

Your type=aor section isn’t doing anything here; it would be very unusual for a provider to register, and it was a bit surprising that they even authenticated.

Although I don’t see how it would cause your problem, I think match=0.0.0.0/0 would be considered bad practice.

16 
<--- Transmitting SIP response (2103 bytes) to TLS:69.171.251.112:59554 --->
SIP/2.0 200 OK
Via: SIP/2.0/TLS 31.13.66.215:5061;rport=59554;received=69.171.251.112;branch=z9hG4bKPj78fbd1ecacabfcf6e4ec2802642545b4;alias
Via: SIP/2.0/TLS [2803:6080:e888:51aa:631d:dd68:146d:a00]:5061;branch=z9hG4bKPj78fbd1ecacabfcf6e4ec2802642545b4
Via: SIP/2.0/TLS [2803:6080:e930:72cf:a69a:c96f:1414:a00]:58585;rport=58585;received=2803:6080:e930:72cf:a69a:c96f:1414:a00;branch=z9hG4bKPjgPpLOtCgE4Ni6dZdAV.9EJV95Ua821dm;alias
Record-Route: <sip:69.171.251.112:59554;transport=TLS;lr>
Record-Route: <sip:onevc-sip-proxy.fbinfra.net:8191;transport=tls;lr>
Call-ID: outgoing:wacid.HBgMOTE4OTU3MTQxMDA1FQIAEhggQUM3RkNGNEM0NjMwQzAxNjA5ODA3MjlFRUY4RjA1NEMcGAw5NzE1ODUxMDM0NzIVAgAVGgA
From: "918957896547" <sip:+918957896547@wa.meta.vc>;tag=dc483986-a50c-40a0-9948-955351ddf373
To: <sip:+971585789632@developer.pronnel.com>;tag=46ba8089-1844-43c3-9170-2c1a551cb2a6
CSeq: 21261 INVITE
Server: Asterisk PBX 22.5.0
Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, INFO, REFER
Contact: <sip:971585789632@20.196.35.162:5061;transport=TLS>
Supported: 100rel, timer, replaces, norefersub
Content-Type: application/sdp
Content-Length:   900

v=0
o=- 1759811125065 4 IN IP4 20.196.35.162
s=Asterisk
c=IN IP4 20.196.35.162
t=0 0
a=msid-semantic:WMS *
a=group:BUNDLE audio
m=audio 12388 UDP/TLS/RTP/SAVPF 111 126
a=connection:new
a=setup:active
a=fingerprint:SHA-256 04:17:0C:BB:B9:D2:DD:7F:36:1C:2C:D0:8B:48:A6:7B:47:1F:F5:AD:C1:67:B9:E0:39:D3:57:BE:6E:B9:59:88
a=ice-ufrag:2548be362343f98050ebb5051326ea7a
a=ice-pwd:1608b56c1d57982d7613f2794c2127d1
a=candidate:H14c41a9a 1 UDP 2130706431 20.196.35.162 12388 typ host
a=rtpmap:111 opus/48000/2
a=fmtp:111 maxplaybackrate=16000;sprop-maxcapturerate=16000;maxaveragebitrate=20000;useinbandfec=1
a=ptime:20
a=maxptime:20
a=sendrecv
a=rtcp-mux
a=ssrc:837099279 cname:d0a73bf4-358c-416c-bb9a-96850ad88cc5
a=msid:4c7cc2dc-547a-49d6-a2c3-2b2f4c5e2dd9 c0de8ae1-c5dc-4467-9f16-d194b39c0599
a=rtcp-fb:* transport-cc
a=mid:audio
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16

Still same thing is happening . Although Contact header value is now set to public ip

17

@ShantanuSingh did you try with my shared pjsip.conf, cause i faced same issue by shared pjsip.conf it is working fine for me

18

i am using media_encryption as dtls and webrtc=yes

19

try with sdes it will works or do you want dtls only?

20 
Endpoint:  <Endpoint/CID.....................................>  <State.....>  <Channels.>
    I/OAuth:  <AuthId/UserName...........................................................>
        Aor:  <Aor............................................>  <MaxContact>
      Contact:  <Aor/ContactUri..........................> <Hash....> <Status> <RTT(ms)..>
  Transport:  <TransportId........>  <Type>  <cos>  <tos>  <BindAddress..................>
   Identify:  <Identify/Endpoint.........................................................>
        Match:  <criteria.........................>
    Channel:  <ChannelId......................................>  <State.....>  <Time.....>
        Exten: <DialedExten...........>  CLCID: <ConnectedLineCID.......>
==========================================================================================

 Endpoint:  whatsapp_endpoint                               Not in use    0 of inf
    OutAuth:  whatsapp_endpoint/971545673423
     InAuth:  whatsapp_endpoint/971545673423
        Aor:  whatsapp_endpoint                             1
      Contact:  whatsapp_endpoint/sip:wa.meta.vc      dc13262203 Avail       528.374
  Transport:  transport-tls             tls      0      0  0.0.0.0:5061
   Identify:  whatsapp_endpoint/whatsapp_endpoint
        Match: 163.70.144.192/32
        Match: 2a03:2880:f0a4:1c0:face:b00c:0:6b34/128
       Header: X-FB-External-Domain: wa.meta.vc


 ParameterName                      : ParameterValue
 ===================================================================================================
 100rel                             : yes
 accept_multiple_sdp_answers        : false
 accountcode                        : 
 acl                                : 
 aggregate_mwi                      : true
 allow                              : (opus)
 allow_overlap                      : true
 allow_subscribe                    : true
 allow_transfer                     : true
 allow_unauthenticated_options      : false
 aors                               : whatsapp_endpoint
 asymmetric_rtp_codec               : false
 auth                               : whatsapp_endpoint
 bind_rtp_to_media_address          : false
 bundle                             : true
 call_group                         : 
 callerid                           : <unknown>
 callerid_privacy                   : allowed_not_screened
 callerid_tag                       : 
 codec_prefs_incoming_answer        : prefer:pending, operation:intersect, keep:all, transcode:allow
 codec_prefs_incoming_offer         : prefer:pending, operation:intersect, keep:all, transcode:allow
 codec_prefs_outgoing_answer        : prefer:pending, operation:intersect, keep:all, transcode:allow
 codec_prefs_outgoing_offer         : prefer:pending, operation:union, keep:all, transcode:allow
 connected_line_method              : invite
 contact_acl                        : 
 contact_user                       : 971545673423
 context                            : from-twilio
 cos_audio                          : 0
 cos_video                          : 0
 device_state_busy_at               : 0
 direct_media                       : false
 direct_media_glare_mitigation      : none
 direct_media_method                : invite
 disable_direct_media_on_nat        : false
 dtls_auto_generate_cert            : Yes
 dtls_ca_file                       : 
 dtls_ca_path                       : 
 dtls_cert_file                     : 
 dtls_cipher                        : 
 dtls_fingerprint                   : SHA-256
 dtls_private_key                   : 
 dtls_rekey                         : 0
 dtls_setup                         : actpass
 dtls_verify                        : Yes
 dtmf_mode                          : rfc4733
 fax_detect                         : false
 fax_detect_timeout                 : 0
 follow_early_media_fork            : true
 force_avp                          : false
 force_rport                        : true
 from_domain                        : developer.pronnel.com
 from_user                          : 971545673423
 g726_non_standard                  : false
 geoloc_incoming_call_profile       : 
 geoloc_outgoing_call_profile       : 
 ice_support                        : true
 identify_by                        : auth_username
 ignore_183_without_sdp             : false
 inband_progress                    : false
 incoming_call_offer_pref           : local
 incoming_mwi_mailbox               : 
 language                           : 
 mailboxes                          : 
 max_audio_streams                  : 1
 max_video_streams                  : 1
 media_address                      : 
 media_encryption                   : dtls
 media_encryption_optimistic        : false
 media_use_received_transport       : true
 message_context                    : 
 moh_passthrough                    : false
 moh_suggest                        : default
 mwi_from_user                      : 
 mwi_subscribe_replaces_unsolicited : no
 named_call_group                   : 
 named_pickup_group                 : 
 notify_early_inuse_ringing         : false
 one_touch_recording                : false
 outbound_auth                      : whatsapp_endpoint
 outbound_proxy                     : 
 outgoing_call_offer_pref           : remote_merge
 overlap_context                    : 
 pickup_group                       : 
 preferred_codec_only               : false
 record_off_feature                 : automixmon
 record_on_feature                  : automixmon
 redirect_method                    : user
 refer_blind_progress               : true
 rewrite_contact                    : true
 rpid_immediate                     : false
 rtcp_mux                           : true
 rtp_engine                         : asterisk
 rtp_ipv6                           : true
 rtp_keepalive                      : 0
 rtp_symmetric                      : true
 rtp_timeout                        : 0
 rtp_timeout_hold                   : 0
 sdp_owner                          : -
 sdp_session                        : Asterisk
 security_negotiation               : no
 send_aoc                           : false
 send_connected_line                : no
 send_diversion                     : true
 send_history_info                  : false
 send_pai                           : false
 send_rpid                          : false
 set_var                            : 
 srtp_tag_32                        : false
 stir_shaken                        : no
 stir_shaken_profile                : 
 sub_min_expiry                     : 0
 subscribe_context                  : 
 suppress_moh_on_sendonly           : false
 suppress_q850_reason_headers       : false
 t38_bind_udptl_to_media_address    : false
 t38_udptl                          : false
 t38_udptl_ec                       : none
 t38_udptl_ipv6                     : false
 t38_udptl_maxdatagram              : 0
 t38_udptl_nat                      : false
 tenantid                           : 
 timers                             : yes
 timers_min_se                      : 90
 timers_sess_expires                : 1800
 tone_zone                          : 
 tos_audio                          : 0
 tos_video                          : 0
 transport                          : transport-tls
 trust_connected_line               : yes
 trust_id_inbound                   : true
 trust_id_outbound                  : false
 use_avpf                           : true
 use_ptime                          : true
 user_eq_phone                      : false
 voicemail_extension                : 
 webrtc                             : yes`Preformatted text`

This is my dtls config for the endpoint

21

I will try once with sdes also