Assistance Required: Asterisk SIP Configuration Issue

Mehwish · January 21, 2025, 11:09am

Hi Team,

I am encountering an issue while generating a call, and the following error is logged in the Asterisk console:

[Jan 21 10:58:24] NOTICE[2258885][C-0000002e]: chan_sip.c:10501 process_sdp: Received SAVPF profile in audio offer but AVPF is not enabled, enabling: audio 55310 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
[Jan 21 10:58:24] WARNING[2258885][C-0000002e]: chan_sip.c:10908 process_sdp: Rejecting secure audio stream without encryption details: audio 55310 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126

Configuration Details:

[web_1111]
type = friend
username = web_1111
secret = 1111
host = dynamic
dtmfmode = rfc2833
canreinvite = no
keepalive = yes
nat = force_rport,comedia
qualify = yes
permit = 0.0.0.0/0.0.0.0
callerid = 1111
voicemail = 1111@ib_user_1
dial = SIP/web_1111
accountcode = 1
setvar = usr_id=1
setvar = extension_id=1
setvar = plan_id=3
callcounter = yes
encryption = yes
directmedia = no
transport = tcp,udp,ws,wss
rtcp_mux = yes
icesupport = yes
avpf = yes
dtlsenable = yes
dtlsverify = fingerprint
dtlscertfile = /etc/asterisk/keys/cert.pem
dtlsprivatekey = /etc/asterisk/keys/privkey.pem
dtlssetup = actpass
disallow = all
allow = alaw,ulaw,g722,vp8
context = ib_user_1
accept_outofcall_message = yes
outofcall_message_context = ib_user_1_message_inbound
auth_message_requests = no

Additional Observations from Browser Console Logs:

INVITE sip:923047120820@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS …
To: sip:923047120820@rhel8.ictbroadcast.com
From: sip:1111@rhel8.ictbroadcast.com
…
a=setup:actpass
…
SIP/2.0 488 Not acceptable here
Server: Asterisk PBX 18.12.1
…

The session is ultimately terminated with the error code 488 Not Acceptable Here.

Request:

Could you help me identify why the secure audio stream is being rejected?
Is there an issue with the AVPF profile or DTLS configuration in my SIP settings?
Any insights on resolving the 488 Not Acceptable Here error?

Thank you for your assistance!

jcolp · January 21, 2025, 11:15am

According to the SIP INVITE that would match as “1111” and not “web_1111”, likely resulting in the wrong configuration being used - specifically a non-WebRTC configuration.

Mehwish · January 22, 2025, 5:24am

i have changes it to web_1111 this error comes
[Jan 22 05:22:08] NOTICE[2498846][C-0000036b]: chan_sip.c:19648 send_check_user_failure_response: Failed to authenticate device sip:web_1111@rhel8.ictbroadcast.com;tag=fl3muqospf for INVITE, code = -1

Mehwish · January 22, 2025, 5:26am

// Options for SimpleUser
const options = {
aor: “sip:web_” + config.extension + “@” + config.hostname, // Use the extension as the caller (AOR - Address of Record)
media: {
constraints: { audio: true, video: false }, // Audio-only call
remote: { audio: getAudioElement(“remoteAudio”) } // Play remote audio in the element with id ‘remoteAudio’
}
};
const server = “wss://” + config.hostname + “:” + config.port + “/ws”; // WebSocket server URL

console.log(options);
console.log(server);
console.log(SIP); // Check if SIP is defined

// Construct the SimpleUser instance
simpleUser = new SIP.Web.SimpleUser(server, options);
// Connect to the server, but do not dial a call immediately
simpleUser.connect()
.then(() => {
console.log(‘SIP connected’); // Only indicate that SIP is connected
// No call initiation here, just waiting for further actions
})
.catch((error) => {
console.error(‘Connection failed:’, error);
});
simpleUser.call(sip:${number}@${sipConfig.hostname}, {
mediaConstraints: { audio: true, video: false }
});

jcolp · January 22, 2025, 9:21am

I would assume since it could previously authenticate and now it can’t, that the credentials are different and weren’t changed.

Mehwish · January 23, 2025, 10:16am

I am using SIP.js (v0.21.2) to implement SIP-based calls in a web application. My code initiates a call, but it fails with a 401 Unauthorized response after sending the initial INVITE. Below is the sequence of events and relevant details:

Code Summary:
Here is the JavaScript code I am using to connect and make the call:

// Dial functionality
dialButton.addEventListener(“click”, () => {
navigator.mediaDevices.getUserMedia({ audio: true })
.then(() => simpleUser.call(sip:${number}@${config.hostname}, {
mediaConstraints: { audio: true, video: false }
}))
.then(() => console.log(“Call initiated”))
.catch((error) => console.error(“Call failed:”, error));
});

[1111]
type=friend
username=1111
secret=1111
host=dynamic
context=from-internal
transport=ws

[from-internal]
exten => _X.,1,Dial(SIP/${EXTEN},30)
exten => _X.,n,Hangup()

Logs:
Here is the relevant part of the logs:

401 Unauthorized response received after sending the INVITE.
The WWW-Authenticate header includes the correct realm and nonce.

Example logs:

INVITE sip:923047120820@rhel8.ictbroadcast.com SIP/2.0
Authorization: Digest algorithm=MD5, username="1111", realm="asterisk", nonce="2477ec76", uri="sip:923047120820@rhel8.ictbroadcast.com"

Environment Details:
    Server: Asterisk PBX 18.12.1
    Browser: Chrome (latest)
    SIP.js Version: 0.21.2
    Transport: WebSocket (wss)

Problem Description:
    The INVITE is sent correctly, but the server responds with 401 Unauthorized.
    The Authorization header in the second INVITE seems to match the WWW-Authenticate challenge.

Questions:
    Why am I still receiving a 401 Unauthorized despite the correct credentials?
    Is there anything I need to adjust in the configuration or SIP.js to address this issue?

Any insights or suggestions would be greatly appreciated.

Tags:
SIP.js WebRTC Asterisk 401 Unauthorized JavaScript

This format should be clear and effective for forums like Stack Overflow or GitHub Issues.

Mehwish · January 23, 2025, 10:19am

when i place a call then it gives error
[1111]
type=friend
username=1111
secret=1111
host=dynamic
context=from-internal
transport=ws

[from-internal]
exten => _X.,1,Dial(SIP/${EXTEN},30)
exten => _X.,n,Hangup()

asterisk log

ib-php8*CLI>

<— SIP read from WS:182.191.65.62:45668 —>
INVITE sip:923047120820@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS v8k3o4ov3cr9.invalid;branch=z9hG4bK5494824
To: sip:923047120820@rhel8.ictbroadcast.com
From: sip:1111@rhel8.ictbroadcast.com;tag=pan1arfv4u
CSeq: 1 INVITE
Call-ID: 7tjm0m0i7rshvmq7q223
Max-Forwards: 70
Contact: sip:cqhbuobb@v8k3o4ov3cr9.invalid;transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Type: application/sdp
Content-Length: 1682

v=0
o=- 1125374324765838027 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 6a300980-796f-453b-88d1-c2dbd0982344
m=audio 51763 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 182.191.65.62
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2161439645 1 udp 2122260223 192.168.1.42 51763 typ host generation 0 network-id 1 network-cost 10
a=candidate:3224273204 1 udp 1686052607 182.191.65.62 51763 typ srflx raddr 192.168.1.42 rport 51763 generation 0 network-id 1 network-cost 10
a=candidate:4263186693 1 tcp 1518280447 192.168.1.42 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:x/rj
a=ice-pwd:l4fiuduZqAN6otl2KNw01ES+
a=ice-options:trickle
a=fingerprint:sha-256 5A:20:B2:4C:2D:F9:60:CA:90:56:69:E9:E1:6E:4A:72:ED:B0:50:D8:53:60:E0:B5:84:A7:F5:DC:D8:00:62:DD
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:6a300980-796f-453b-88d1-c2dbd0982344 2652e002-76b6-4a08-95de-1b45e57e2591
a=rtcp-mux
a=rtcp-rsize
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:1419390669 cname:KdSSVpaeeecPWuJS
a=ssrc:1419390669 msid:6a300980-796f-453b-88d1-c2dbd0982344 2652e002-76b6-4a08-95de-1b45e57e2591
<------------->
— (13 headers 40 lines) —
Using INVITE request as basis request - 7tjm0m0i7rshvmq7q223
Found peer ‘1111’ for ‘1111’ from 182.191.65.62:45668

<— Reliably Transmitting (no NAT) to 182.191.65.62:5060 —>
SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS v8k3o4ov3cr9.invalid;branch=z9hG4bK5494824;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=pan1arfv4u
To: sip:923047120820@rhel8.ictbroadcast.com;tag=as39b5be0d
Call-ID: 7tjm0m0i7rshvmq7q223
CSeq: 1 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“52899610”
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7tjm0m0i7rshvmq7q223’ in 32000 ms (Method: INVITE)

<— SIP read from WS:182.191.65.62:45668 —>
ACK sip:923047120820@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS v8k3o4ov3cr9.invalid;branch=z9hG4bK5494824
To: sip:923047120820@rhel8.ictbroadcast.com;tag=as39b5be0d
From: sip:1111@rhel8.ictbroadcast.com;tag=pan1arfv4u
Call-ID: 7tjm0m0i7rshvmq7q223
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

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

<— SIP read from WS:182.191.65.62:45668 —>
INVITE sip:923047120820@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS v8k3o4ov3cr9.invalid;branch=z9hG4bK7978295
To: sip:923047120820@rhel8.ictbroadcast.com
From: sip:1111@rhel8.ictbroadcast.com;tag=pan1arfv4u
CSeq: 2 INVITE
Call-ID: 7tjm0m0i7rshvmq7q223
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“1111”, realm=“asterisk”, nonce=“52899610”, uri="sip:923047120820@rhel8.ictbroadcast.com", response=“1e146397df7acf8638553558a3c019dc”
Contact: sip:cqhbuobb@v8k3o4ov3cr9.invalid;transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Type: application/sdp
Content-Length: 1682

v=0
o=- 1125374324765838027 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS 6a300980-796f-453b-88d1-c2dbd0982344
m=audio 51763 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 182.191.65.62
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:2161439645 1 udp 2122260223 192.168.1.42 51763 typ host generation 0 network-id 1 network-cost 10
a=candidate:3224273204 1 udp 1686052607 182.191.65.62 51763 typ srflx raddr 192.168.1.42 rport 51763 generation 0 network-id 1 network-cost 10
a=candidate:4263186693 1 tcp 1518280447 192.168.1.42 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:x/rj
a=ice-pwd:l4fiuduZqAN6otl2KNw01ES+
a=ice-options:trickle
a=fingerprint:sha-256 5A:20:B2:4C:2D:F9:60:CA:90:56:69:E9:E1:6E:4A:72:ED:B0:50:D8:53:60:E0:B5:84:A7:F5:DC:D8:00:62:DD
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:6a300980-796f-453b-88d1-c2dbd0982344 2652e002-76b6-4a08-95de-1b45e57e2591
a=rtcp-mux
a=rtcp-rsize
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:1419390669 cname:KdSSVpaeeecPWuJS
a=ssrc:1419390669 msid:6a300980-796f-453b-88d1-c2dbd0982344 2652e002-76b6-4a08-95de-1b45e57e2591
<------------->
— (14 headers 40 lines) —
Using INVITE request as basis request - 7tjm0m0i7rshvmq7q223
Found peer ‘1111’ for ‘1111’ from 182.191.65.62:45668
== Using SIP RTP CoS mark 5
Got SDP version 2 and unique parts [- 1125374324765838027 IN IP4 127.0.0.1]
[Jan 23 09:58:42] NOTICE[2880513][C-00000001]: chan_sip.c:10501 process_sdp: Received SAVPF profle in audio offer but AVPF is not enabled, enabling: audio 51763 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
Found RTP audio format 111
Found RTP audio format 63
Found RTP audio format 9
Found RTP audio format 0
Found RTP audio format 8
Found RTP audio format 13
Found RTP audio format 110
Found RTP audio format 126
Found audio description format opus for ID 111
Found unknown media description format red for ID 63
Found audio description format G722 for ID 9
Found audio description format PCMU for ID 0
Found audio description format PCMA for ID 8
Found audio description format CN for ID 13
Found unknown media description format telephone-event for ID 110
Found audio description format telephone-event for ID 126
[Jan 23 09:58:42] WARNING[2880513][C-00000001]: chan_sip.c:10908 process_sdp: Rejecting secure audio stream without encryption details: audio 51763 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126

<— Reliably Transmitting (no NAT) to 182.191.65.62:5060 —>
SIP/2.0 488 Not acceptable here
Via: SIP/2.0/WSS v8k3o4ov3cr9.invalid;branch=z9hG4bK7978295;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=pan1arfv4u
To: sip:923047120820@rhel8.ictbroadcast.com;tag=as39b5be0d
Call-ID: 7tjm0m0i7rshvmq7q223
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
Scheduling destruction of SIP dialog ‘7tjm0m0i7rshvmq7q223’ in 32000 ms (Method: INVITE)
ib-php8*CLI>

<— SIP read from WS:182.191.65.62:45668 —>
ACK sip:923047120820@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS v8k3o4ov3cr9.invalid;branch=z9hG4bK7978295
To: sip:923047120820@rhel8.ictbroadcast.com;tag=as39b5be0d
From: sip:1111@rhel8.ictbroadcast.com;tag=pan1arfv4u
Call-ID: 7tjm0m0i7rshvmq7q223
CSeq: 2 ACK
Max-Forwards: 70
Content-Length: 0

<------------->
— (8 headers 0 lines) —
ib-php8*CLI>

david551 · January 23, 2025, 10:55am

The error is fairly clear from this message.

Also note that chan_sip is no longer supported, and was not included at all in the last two versions of Asterisk. Asterisk 18 is in security fixes only status.

401 is not an error. Your first posting cut off the response to the 401, making it appear that the client was unable to handle it.

Siperb · January 23, 2025, 11:03am

This is not enough for WebRTC to work on old Asterisk boxes.

transport=wss
allow=opus,ulaw,vp9,vp8,h264
encryption=yes
avpf=yes
force_avp=yes
icesupport=yes
rtcp_mux=yes
dtlsenable=yes
dtlsverify=fingerprint
dtlscertfile=/home/pi/certs/cert.pem
dtlscafile=/home/pi/ca/CA-cert.crt
dtlssetup=actpass

You are in for quite a bit of difficulty getting this to work tho. If upgrading is not an option, contact me privately.

Mehwish · January 24, 2025, 6:12am

when i generate a call it hangup why?

<------------>
– Executing [12132942943@from-internal:1] Dial(“SIP/1111-00000004”, “SIP/12132942943@Telnyx,30”) in new stack
== Using SIP RTP CoS mark 5
Audio is at 18600
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.76.120.10:5060:
INVITE sip:12132942943@sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 216.128.136.88:5060;branch=z9hG4bK653f00ed
Max-Forwards: 70
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com
Contact: sip:12132942876@216.128.136.88:5060
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX 18.12.1
Date: Fri, 24 Jan 2025 06:05:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 1650102339 1650102339 IN IP4 216.128.136.88
s=Asterisk PBX 18.12.1
c=IN IP4 216.128.136.88
t=0 0
m=audio 18600 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

-- Called SIP/12132942943@Telnyx

<— SIP read from UDP:192.76.120.10:5060 —>
SIP/2.0 100 Telnyx Trying
Via: SIP/2.0/UDP 216.128.136.88:5060;branch=z9hG4bK653f00ed;rport=5060
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 102 INVITE
Server: Telnyx SIP Proxy
Content-Length: 0

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

<— SIP read from UDP:192.76.120.10:5060 —>
SIP/2.0 407 Proxy Authentication Required
Via: SIP/2.0/UDP 216.128.136.88:5060;rport=5060;branch=z9hG4bK653f00ed
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com;tag=00rUKaa850jpg
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 102 INVITE
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path
Allow-Events: talk, hold, conference, refer
Proxy-Authenticate: Digest realm=“sip.telnyx.com”, nonce=“f58317c1-ae92-4483-991a-220f522f57f6”, algorithm=MD5, qop=“auth”, opaque=“1249eb00-6850-45b1-b624-23792ac1fa04/10.239.197.24”
Content-Length: 0

<------------->
— (12 headers 0 lines) —
Transmitting (no NAT) to 192.76.120.10:5060:
ACK sip:12132942943@sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 216.128.136.88:5060;branch=z9hG4bK653f00ed
Max-Forwards: 70
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com;tag=00rUKaa850jpg
Contact: sip:12132942876@216.128.136.88:5060
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 102 ACK
User-Agent: Asterisk PBX 18.12.1
Content-Length: 0

Audio is at 18600
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec gsm to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (no NAT) to 192.76.120.10:5060:
INVITE sip:12132942943@sip.telnyx.com SIP/2.0
Via: SIP/2.0/UDP 216.128.136.88:5060;branch=z9hG4bK657c6e5b
Max-Forwards: 70
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com
Contact: sip:12132942876@216.128.136.88:5060
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 103 INVITE
User-Agent: Asterisk PBX 18.12.1
Proxy-Authorization: Digest username=“Danish7180”, realm=“sip.telnyx.com”, algorithm=MD5, uri="sip:12132942943@sip.telnyx.com", nonce=“f58317c1-ae92-4483-991a-220f522f57f6”, response=“d3d9a53d6d60fb736c1941041df2911b”, opaque=“1249eb00-6850-45b1-b624-23792ac1fa04/10.239.197.24”, qop=auth, cnonce=“059aec6f”, nc=00000001
Date: Fri, 24 Jan 2025 06:05:40 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Type: application/sdp
Content-Length: 291

v=0
o=root 1650102339 1650102340 IN IP4 216.128.136.88
s=Asterisk PBX 18.12.1
c=IN IP4 216.128.136.88
t=0 0
m=audio 18600 RTP/AVP 0 8 3 101
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=maxptime:150
a=sendrecv

<— SIP read from UDP:192.76.120.10:5060 —>
SIP/2.0 100 Telnyx Trying
Via: SIP/2.0/UDP 216.128.136.88:5060;branch=z9hG4bK657c6e5b;rport=5060
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 103 INVITE
Server: Telnyx SIP Proxy
Content-Length: 0

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

<— SIP read from UDP:192.76.120.10:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/UDP 216.128.136.88:5060;rport=5060;branch=z9hG4bK657c6e5b
Record-Route: sip:10.255.0.1;r2=on;lr;ftag=as2619a22b
Record-Route: sip:192.76.120.10;r2=on;lr;ftag=as2619a22b
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com;tag=19HmN5tB3988B
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 103 INVITE
Contact: sip:12132942943@10.239.197.24:5070;transport=udp
Accept: application/sdp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path
Allow-Events: talk, hold, conference, refer
Content-Length: 0

<------------->
— (14 headers 0 lines) —
sip_route_dump: route/path hop: sip:192.76.120.10;r2=on;lr;ftag=as2619a22b
sip_route_dump: route/path hop: sip:10.255.0.1;r2=on;lr;ftag=as2619a22b

<— SIP read from UDP:192.76.120.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.128.136.88:5060;rport=5060;branch=z9hG4bK657c6e5b
Record-Route: sip:10.255.0.1;r2=on;lr;ftag=as2619a22b
Record-Route: sip:192.76.120.10;r2=on;lr;ftag=as2619a22b
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com;tag=19HmN5tB3988B
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 103 INVITE
Contact: sip:12132942943@10.239.197.24:5070;transport=udp
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path
Allow-Events: talk, hold, conference, refer
Content-Type: application/sdp
Content-Disposition: session
Content-Length: 309

v=0
o=FreeSWITCH 1737667984 1737667985 IN IP4 50.114.147.12
s=FreeSWITCH
c=IN IP4 50.114.147.12
t=0 0
m=audio 30756 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=sendrecv
a=silenceSupp:off - - - -
a=ptime:20
a=mid:audio
a=rtcp:30757 IN IP4 50.114.147.12
<------------->
— (15 headers 14 lines) —
Got SDP version 1737667985 and unique parts [FreeSWITCH 1737667984 IN IP4 50.114.147.12]
Found RTP audio format 0
Found RTP audio format 101
Found audio description format PCMU for ID 0
Found audio description format telephone-event for ID 101
Capabilities: us - (ulaw|alaw|gsm|h263), peer - audio=(ulaw)/video=(nothing)/text=(nothing), combined - (ulaw)
Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|)
> 0x7fae40008420 – Strict RTP learning after remote address set to: 50.114.147.12:30756
Peer audio RTP is at port 50.114.147.12:30756
sip_route_dump: route/path hop: sip:192.76.120.10;r2=on;lr;ftag=as2619a22b
sip_route_dump: route/path hop: sip:10.255.0.1;r2=on;lr;ftag=as2619a22b
set_destination: Parsing sip:192.76.120.10;r2=on;lr;ftag=as2619a22b for address/port to send to
set_destination: set destination to 192.76.120.10:5060
Transmitting (no NAT) to 192.76.120.10:5060:
ACK sip:12132942943@10.239.197.24:5070;transport=udp SIP/2.0
Via: SIP/2.0/UDP 216.128.136.88:5060;branch=z9hG4bK4213c389
Route: sip:192.76.120.10;r2=on;lr;ftag=as2619a22b,sip:10.255.0.1;r2=on;lr;ftag=as2619a22b
Max-Forwards: 70
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com;tag=19HmN5tB3988B
Contact: sip:12132942876@216.128.136.88:5060
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 103 ACK
User-Agent: Asterisk PBX 18.12.1
Content-Length: 0

-- SIP/Telnyx-00000005 is ringing

<— Transmitting (no NAT) to 182.191.65.62:5060 —>
SIP/2.0 180 Ringing
Via: SIP/2.0/WSS g8kflshtkftn.invalid;branch=z9hG4bK9586839;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=042s25glmc
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as2f309896
Call-ID: nvbh96tms8hkmlkli1m4
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:12132942943@216.128.136.88:5060;transport=ws
Content-Length: 0

<------------>
– SIP/Telnyx-00000005 answered SIP/1111-00000004
Audio is at 13504
Adding codec ulaw to SDP
Adding codec alaw to SDP
Adding codec opus to SDP
Adding non-codec 0x1 (telephone-event) to SDP

<— Reliably Transmitting (no NAT) to 182.191.65.62:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS g8kflshtkftn.invalid;branch=z9hG4bK9586839;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=042s25glmc
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as2f309896
Call-ID: nvbh96tms8hkmlkli1m4
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:12132942943@216.128.136.88:5060;transport=ws
Content-Type: application/sdp
Content-Length: 600

v=0
o=root 1530018043 1530018043 IN IP4 216.128.136.88
s=Asterisk PBX 18.12.1
c=IN IP4 216.128.136.88
t=0 0
m=audio 13504 RTP/SAVPF 0 8 111 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 opus/48000/2
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:60
a=ice-ufrag:3984b44279d2289a1548e03f6cab0f7c
a=ice-pwd:7f3be6f82e5986163e3f89951ebc5740
a=candidate:Hd8808858 1 UDP 2130706431 216.128.136.88 13504 typ host
a=candidate:Hd8808858 2 UDP 2130706430 216.128.136.88 13505 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256
a=rtcp-mux
a=sendrecv

<------------>
– Channel SIP/Telnyx-00000005 joined ‘simple_bridge’ basic-bridge <2281bb09-cfe8-45a2-9893-a955fb978d78>
– Channel SIP/1111-00000004 joined ‘simple_bridge’ basic-bridge <2281bb09-cfe8-45a2-9893-a955fb978d78>
> 0x7fae40008420 – Strict RTP switching to RTP target address 50.114.147.12:30756 as source

<— SIP read from WS:182.191.65.62:51714 —>
ACK sip:12132942943@216.128.136.88:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS g8kflshtkftn.invalid;branch=z9hG4bK5671750
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as2f309896
From: sip:1111@rhel8.ictbroadcast.com;tag=042s25glmc
CSeq: 2 ACK
Call-ID: nvbh96tms8hkmlkli1m4
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Length: 0

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

<— SIP read from WS:182.191.65.62:51714 —>
BYE sip:12132942943@216.128.136.88:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS g8kflshtkftn.invalid;branch=z9hG4bK7049698
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as2f309896
From: sip:1111@rhel8.ictbroadcast.com;tag=042s25glmc
CSeq: 3 BYE
Call-ID: nvbh96tms8hkmlkli1m4
Max-Forwards: 70
Reason: SIP;cause=488;text=“Not Acceptable Here”
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Length: 0

<------------->
— (11 headers 0 lines) —
Scheduling destruction of SIP dialog ‘nvbh96tms8hkmlkli1m4’ in 32000 ms (Method: BYE)

<— Transmitting (no NAT) to 182.191.65.62:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/WSS g8kflshtkftn.invalid;branch=z9hG4bK7049698;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=042s25glmc
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as2f309896
Call-ID: nvbh96tms8hkmlkli1m4
CSeq: 3 BYE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

<------------>
– Channel SIP/1111-00000004 left ‘simple_bridge’ basic-bridge <2281bb09-cfe8-45a2-9893-a955fb978d78>
== Spawn extension (from-internal, 12132942943, 1) exited non-zero on ‘SIP/1111-00000004’
– Channel SIP/Telnyx-00000005 left ‘simple_bridge’ basic-bridge <2281bb09-cfe8-45a2-9893-a955fb978d78>
Scheduling destruction of SIP dialog ‘549c0eb461efc02449909031032bdd76@216.128.136.88:5060’ in 32000 ms (Method: INVITE)
set_destination: Parsing sip:192.76.120.10;r2=on;lr;ftag=as2619a22b for address/port to send to
set_destination: set destination to 192.76.120.10:5060
Reliably Transmitting (no NAT) to 192.76.120.10:5060:
BYE sip:12132942943@10.239.197.24:5070;transport=udp SIP/2.0
Via: SIP/2.0/UDP 216.128.136.88:5060;branch=z9hG4bK369c3e01
Route: sip:192.76.120.10;r2=on;lr;ftag=as2619a22b,sip:10.255.0.1;r2=on;lr;ftag=as2619a22b
Max-Forwards: 70
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com;tag=19HmN5tB3988B
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 104 BYE
User-Agent: Asterisk PBX 18.12.1
Proxy-Authorization: Digest username=“Danish7180”, realm=“sip.telnyx.com”, algorithm=MD5, uri=“sip:12132942943@10.239.197.24:5070”, nonce=“f58317c1-ae92-4483-991a-220f522f57f6”, response=“4196bb64e5dd86821649453b85af8417”, opaque=“1249eb00-6850-45b1-b624-23792ac1fa04/10.239.197.24”, qop=auth, cnonce=“4ed79404”, nc=00000002
X-Asterisk-HangupCause: Normal Clearing
X-Asterisk-HangupCauseCode: 16
Content-Length: 0

<— SIP read from UDP:192.76.120.10:5060 —>
SIP/2.0 200 OK
Via: SIP/2.0/UDP 216.128.136.88:5060;rport=5060;branch=z9hG4bK369c3e01
From: sip:12132942876@216.128.136.88;tag=as2619a22b
To: sip:12132942943@sip.telnyx.com;tag=19HmN5tB3988B
Call-ID: 549c0eb461efc02449909031032bdd76@216.128.136.88:5060
CSeq: 104 BYE
Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REFER, NOTIFY
Supported: path
Content-Length: 0

<------------->
— (9 headers 0 lines) —
Really destroying SIP dialog ‘549c0eb461efc02449909031032bdd76@216.128.136.88:5060’ Method: INVITE
ib-php8*CLI>

Siperb · January 24, 2025, 8:47am

Reason: SIP;cause=488;text=“Not Acceptable Here”

This is from SIP.js the INVITE is not in the format that it accepts, maybe a codec issue, maybe it’s not DTLS. Check the CLI in the browser and the INVITE in the websocket connection raw data.

Mehwish · January 24, 2025, 8:55am

Fri Jan 24 2025 13:52:25 GMT+0500 (Pakistan Standard Time) | sip.Transport | WebSocket opened wss://rhel8.ictbroadcast.com:8089/ws
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:25 GMT+0500 (Pakistan Standard Time) | sip.Transport | Transitioned from Connecting to Connected
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:25 GMT+0500 (Pakistan Standard Time) | sip.SessionManager | Connected
popup.js:130 SIP connected
popup.js:131 0.21.2
popup.js:156 Microphone access granted.
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.SimpleUser | [Anonymous] Beginning Session…
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.SessionManager | Beginning Session…
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Inviter.invite
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Session i4lg6as4bknus1tfr90c44lfs1971u transitioned to state Establishing
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.SessionManager | [i4lg6as4bknus1tfr90c44lfs1971u] Session state changed to Establishing
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Transport | Sending WebSocket message:

INVITE sip:12132942943@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK5965427
To: sip:12132942943@rhel8.ictbroadcast.com
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
CSeq: 1 INVITE
Call-ID: i4lg6as4bknus1tfr90c
Max-Forwards: 70
Contact: sip:bgunh472@52a7g871st95.invalid;transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Type: application/sdp
Content-Length: 1680

v=0
o=- 2719051949487533909 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS d361a0b2-0c4c-4925-b3a4-06139aa324a9
m=audio 58728 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 182.191.65.62
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1454291960 1 udp 2122260223 192.168.1.42 58728 typ host generation 0 network-id 1 network-cost 10
a=candidate:374655313 1 udp 1686052607 182.191.65.62 58728 typ srflx raddr 192.168.1.42 rport 58728 generation 0 network-id 1 network-cost 10
a=candidate:677468512 1 tcp 1518280447 192.168.1.42 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:UOvV
a=ice-pwd:5zVoqy98Eepo0gjTrWoz/Rm+
a=ice-options:trickle
a=fingerprint:sha-256 95:C2:CC:9A:48:AA:92:A6:C0:77:25:EF:94:FE:2A:E8:DC:C8:CD:B3:0B:FC:B0:3D:10:A8:04:EF:7C:5B:39:3A
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:d361a0b2-0c4c-4925-b3a4-06139aa324a9 7e8ec24a-78a3-4549-9c5b-3b48d46c3342
a=rtcp-mux
a=rtcp-rsize
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:3266618501 cname:uo58Cv0QMXbgg7+Z
a=ssrc:3266618501 msid:d361a0b2-0c4c-4925-b3a4-06139aa324a9 7e8ec24a-78a3-4549-9c5b-3b48d46c3342

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.SessionManager | [i4lg6as4bknus1tfr90c44lfs1971u] Sent INVITE
popup.js:164 Call initiated successfully.
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 401 Unauthorized
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK5965427;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as511a45fc
Call-ID: i4lg6as4bknus1tfr90c
CSeq: 1 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
WWW-Authenticate: Digest algorithm=MD5, realm=“asterisk”, nonce=“557b2c3f”
Content-Length: 0

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Transport | Sending WebSocket message:

ACK sip:12132942943@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK5965427
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as511a45fc
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
Call-ID: i4lg6as4bknus1tfr90c
CSeq: 1 ACK
Max-Forwards: 70
Content-Length: 0

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Transport | Sending WebSocket message:

INVITE sip:12132942943@rhel8.ictbroadcast.com SIP/2.0
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK3310183
To: sip:12132942943@rhel8.ictbroadcast.com
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
CSeq: 2 INVITE
Call-ID: i4lg6as4bknus1tfr90c
Max-Forwards: 70
Authorization: Digest algorithm=MD5, username=“1111”, realm=“asterisk”, nonce=“557b2c3f”, uri="sip:12132942943@rhel8.ictbroadcast.com", response=“984112fcacc765eb0390a134a74e6b33”
Contact: sip:bgunh472@52a7g871st95.invalid;transport=ws;ob
Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Type: application/sdp
Content-Length: 1680

v=0
o=- 2719051949487533909 2 IN IP4 127.0.0.1
s=-
t=0 0
a=group:BUNDLE 0
a=extmap-allow-mixed
a=msid-semantic: WMS d361a0b2-0c4c-4925-b3a4-06139aa324a9
m=audio 58728 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126
c=IN IP4 182.191.65.62
a=rtcp:9 IN IP4 0.0.0.0
a=candidate:1454291960 1 udp 2122260223 192.168.1.42 58728 typ host generation 0 network-id 1 network-cost 10
a=candidate:374655313 1 udp 1686052607 182.191.65.62 58728 typ srflx raddr 192.168.1.42 rport 58728 generation 0 network-id 1 network-cost 10
a=candidate:677468512 1 tcp 1518280447 192.168.1.42 9 typ host tcptype active generation 0 network-id 1 network-cost 10
a=ice-ufrag:UOvV
a=ice-pwd:5zVoqy98Eepo0gjTrWoz/Rm+
a=ice-options:trickle
a=fingerprint:sha-256 95:C2:CC:9A:48:AA:92:A6:C0:77:25:EF:94:FE:2A:E8:DC:C8:CD:B3:0B:FC:B0:3D:10:A8:04:EF:7C:5B:39:3A
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:d361a0b2-0c4c-4925-b3a4-06139aa324a9 7e8ec24a-78a3-4549-9c5b-3b48d46c3342
a=rtcp-mux
a=rtcp-rsize
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:3266618501 cname:uo58Cv0QMXbgg7+Z
a=ssrc:3266618501 msid:d361a0b2-0c4c-4925-b3a4-06139aa324a9 7e8ec24a-78a3-4549-9c5b-3b48d46c3342

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 100 Trying
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK3310183;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
To: sip:12132942943@rhel8.ictbroadcast.com
Call-ID: i4lg6as4bknus1tfr90c
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:12132942943@216.128.136.88:5060;transport=ws
Content-Length: 0

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:32 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Inviter.onTrying
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 180 Ringing
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK3310183;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as04a30f54
Call-ID: i4lg6as4bknus1tfr90c
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:12132942943@216.128.136.88:5060;transport=ws
Content-Length: 0

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.invite-dialog | INVITE dialog i4lg6as4bknus1tfr90c44lfs1971uas04a30f54 constructed
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Inviter.onProgress
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK3310183;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as04a30f54
Call-ID: i4lg6as4bknus1tfr90c
CSeq: 2 INVITE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Contact: sip:12132942943@216.128.136.88:5060;transport=ws
Content-Type: application/sdp
Content-Length: 596

v=0
o=root 37344758 37344758 IN IP4 216.128.136.88
s=Asterisk PBX 18.12.1
c=IN IP4 216.128.136.88
t=0 0
m=audio 19628 RTP/SAVPF 0 8 111 126
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:111 opus/48000/2
a=rtpmap:126 telephone-event/8000
a=fmtp:126 0-16
a=maxptime:60
a=ice-ufrag:233095dc1fd8a6dd160944f83e15d611
a=ice-pwd:6a1399364586e53e7b04cd994fdff555
a=candidate:Hd8808858 1 UDP 2130706431 216.128.136.88 19628 typ host
a=candidate:Hd8808858 2 UDP 2130706430 216.128.136.88 19629 typ host
a=connection:new
a=setup:active
a=fingerprint:SHA-256
a=rtcp-mux
a=sendrecv

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Inviter.onAccept
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.SessionDescriptionHandler | SessionDescriptionHandler.setDescription failed - OperationError: Failed to execute ‘setRemoteDescription’ on ‘RTCPeerConnection’: Failed to parse SessionDescription. a=fingerprint:SHA-256 Failed to create fingerprint from the digest.
print @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
error @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2
Promise.catch
setDescription @ sip-0.21.2.min.js:2
setAnswer @ sip-0.21.2.min.js:2
onAccept @ sip-0.21.2.min.js:2
onAccept @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponseFromTransport @ sip-0.21.2.min.js:2
receiveIncomingResponseFromTransport @ sip-0.21.2.min.js:2
onTransportMessage @ sip-0.21.2.min.js:2
transport.onMessage @ sip-0.21.2.min.js:2
onWebSocketMessage @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2Understand this errorAI
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Session.setAnswer: SDH setDescription rejected…
print @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
error @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2
Promise.catch
setAnswer @ sip-0.21.2.min.js:2
onAccept @ sip-0.21.2.min.js:2
onAccept @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponseFromTransport @ sip-0.21.2.min.js:2
receiveIncomingResponseFromTransport @ sip-0.21.2.min.js:2
onTransportMessage @ sip-0.21.2.min.js:2
transport.onMessage @ sip-0.21.2.min.js:2
onWebSocketMessage @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2Understand this errorAI
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Failed to execute ‘setRemoteDescription’ on ‘RTCPeerConnection’: Failed to parse SessionDescription. a=fingerprint:SHA-256 Failed to create fingerprint from the digest.
print @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
error @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2
Promise.catch
setAnswer @ sip-0.21.2.min.js:2
onAccept @ sip-0.21.2.min.js:2
onAccept @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponseFromTransport @ sip-0.21.2.min.js:2
receiveIncomingResponseFromTransport @ sip-0.21.2.min.js:2
onTransportMessage @ sip-0.21.2.min.js:2
transport.onMessage @ sip-0.21.2.min.js:2
onWebSocketMessage @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2Understand this errorAI
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Failed to execute ‘setRemoteDescription’ on ‘RTCPeerConnection’: Failed to parse SessionDescription. a=fingerprint:SHA-256 Failed to create fingerprint from the digest.
print @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
genericLog @ sip-0.21.2.min.js:2
error @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2
Promise.catch
onAccept @ sip-0.21.2.min.js:2
onAccept @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponse @ sip-0.21.2.min.js:2
receiveResponseFromTransport @ sip-0.21.2.min.js:2
receiveIncomingResponseFromTransport @ sip-0.21.2.min.js:2
onTransportMessage @ sip-0.21.2.min.js:2
transport.onMessage @ sip-0.21.2.min.js:2
onWebSocketMessage @ sip-0.21.2.min.js:2
(anonymous) @ sip-0.21.2.min.js:2Understand this errorAI
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.invite-dialog | INVITE dialog i4lg6as4bknus1tfr90c44lfs1971uas04a30f54 sending ACK request
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Transport | Sending WebSocket message:

ACK sip:12132942943@216.128.136.88:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK8889438
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as04a30f54
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
CSeq: 2 ACK
Call-ID: i4lg6as4bknus1tfr90c
Max-Forwards: 70
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Length: 0

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.invite-dialog | INVITE dialog i4lg6as4bknus1tfr90c44lfs1971uas04a30f54 sending BYE request
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Transport | Sending WebSocket message:

BYE sip:12132942943@216.128.136.88:5060;transport=ws SIP/2.0
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK5177273
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as04a30f54
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
CSeq: 3 BYE
Call-ID: i4lg6as4bknus1tfr90c
Max-Forwards: 70
Reason: SIP;cause=488;text=“Not Acceptable Here”
Supported: outbound
User-Agent: SIP.js/0.21.2
Content-Length: 0

sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.invite-dialog | INVITE dialog i4lg6as4bknus1tfr90c44lfs1971uas04a30f54 destroyed
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Session i4lg6as4bknus1tfr90c44lfs1971u transitioned to state Terminated
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.SessionManager | [i4lg6as4bknus1tfr90c44lfs1971u] Session state changed to Terminated
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Inviter | Session i4lg6as4bknus1tfr90c44lfs1971u in state Terminated is being disposed
sip-0.21.2.min.js:2 Fri Jan 24 2025 13:52:33 GMT+0500 (Pakistan Standard Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 200 OK
Via: SIP/2.0/WSS 52a7g871st95.invalid;branch=z9hG4bK5177273;received=182.191.65.62
From: sip:1111@rhel8.ictbroadcast.com;tag=44lfs1971u
To: sip:12132942943@rhel8.ictbroadcast.com;tag=as04a30f54
Call-ID: i4lg6as4bknus1tfr90c
CSeq: 3 BYE
Server: Asterisk PBX 18.12.1
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
Supported: replaces, timer
Content-Length: 0

jcolp · January 24, 2025, 9:25am

It rejected the SDP in the 200 OK, because it does not include DTLS-SRTP.

Mehwish · January 24, 2025, 9:26am

Please can you guide me how we can resolve it?

jcolp · January 24, 2025, 9:32am

Someone else may, I don’t do chan_sip anymore.

system · February 23, 2025, 9:32am

This topic was automatically closed 30 days after the last reply. New replies are no longer allowed.

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Assistance Required: Asterisk SIP Configuration Issue

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