Failed to create fingerprint from the digest.

[wagnerdracha]

Hello everyone

First of all! Thanks for sharing this project! It is amazing!

So, I am getting this error message.

I tried to figure out the error, but I am not successful.

What can it be?

Thanks in advanced!

[InnovateAsterisk]

What version of asterisk are you using? It seems like the failure is in the formation of a valid SDP. This could be because asterisk is somehow not creating the SDP according to the webrtc settings. Are you using webrtc=yes in your config or specifying your own settings?

[wagnerdracha]

Asterisk version is 16.7.0

So, are "webrtc=yes" must be on pjsip.conf file?

[wagnerdracha]

My /etc/asterisk/pjsip.conf:

`;--------------------------------------------------------------------------------; ; Do NOT edit this file as it is auto-generated by IssabelPBX. All modifications ; ; to this file must be done via the web gui. There are alternative files to make ; ; custom modifications, details at: http://issabel.org/configuration_files ; ;--------------------------------------------------------------------------------;

include pjsip_transport_custom.conf

include pjsip_transport_additional.conf

include pjsip_custom.conf

include pjsip_additional.conf

[dummy_endpoint] type=endpoint context=dummy-invalid

[global] type=global user_agent=Issabel-4.1 default_outbound_endpoint=dummy_endpoint

include pjsip_custom_post.conf

[webrtc_client] dtls_auto_generate_cert=yes webrtc=yes`

[InnovateAsterisk]

At what point does this happen? Or can you describe the step you took to create this. Also can you go to the network tab on the developer tools, find the websocket connection called /ws/ (you can filter for websocket). Then select it, and once you have done that, you will see a new set of tabs, one with messages. These are the actual sip messages being passed to and from the server (via the secure websocket connection).

I’m looking for the last INVITE with the SDP that causes this.

[wagnerdracha]

Hello @InnovateAsterisk! Thanks for your help!

Follow the answers to your questions

At what point does this happen? Answer: So, this error happen when I click on answer call.

Or can you describe the step you took to create this. Answer: The file /etc/asterisk/pjsip.conf is a issabel default file. I just add this:

[webrtc_client]
dtls_auto_generate_cert=yes
webrtc=yes`

Also can you go to the network tab on the... Answer: Abou this, I don't found it. I am sorry, but I am a new issabel user.

Thanks again!

[InnovateAsterisk]

This Wiki will help with the INVITE message.

https://github.com/InnovateAsterisk/Browser-Phone/wiki/WebRTC-SIP-Trace-Capture

[opera-capital]

Hi @InnovateAsterisk

How are you?

I am sorry to respond one month later and again thanks in advanced for your help!

So, now I understand what you sayed!

I opened the DevTools, filter to get just WS messages and reloaded the page.

After that, I called with a SIP Phone (2103) to Browser-Phone (2106) and I got this messages:

The last INVITE before call ended:

`INVITE sip:ai79apv1@192.0.2.223;transport=wss SIP/2.0 Via: SIP/2.0/WS 192.168.4.240:5060;branch=z9hG4bK706880ae;rport Max-Forwards: 70 From: "Wagner" sip:2103@192.168.4.240;tag=as7529e8f3 To: sip:ai79apv1@192.0.2.223;transport=wss Contact: sip:2103@192.168.4.240:5060;transport=ws Call-ID: 496f762270fbfd091ab2f62f52158e34@192.168.4.240:5060 CSeq: 102 INVITE User-Agent: IPBX-2.11.0(16.7.0) Date: Thu, 07 Dec 2023 18:16:20 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Type: application/sdp Content-Length: 820

v=0 o=root 367670485 367670485 IN IP4 192.168.4.240 s=Asterisk PBX 16.7.0 c=IN IP4 192.168.4.240 t=0 0 m=audio 10792 RTP/SAVPF 0 3 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:150 a=ice-ufrag:5b7204a039d3d34f1905b46045b520e8 a=ice-pwd:6f37591d5a2471135a3fbfb44e6dda5c a=candidate:Hc0a803f0 1 UDP 2130706431 192.168.3.240 10792 typ host a=candidate:Hc0a804f0 1 UDP 2130706431 192.168.4.240 10792 typ host a=candidate:Hc0a803f0 2 UDP 2130706430 192.168.3.240 10793 typ host a=candidate:Hc0a804f0 2 UDP 2130706430 192.168.4.240 10793 typ host a=connection:new a=setup:actpass a=fingerprint:SHA-256 2B:A4:28:2E:18:F1:D6:8A:E4:B4:EA:C0:70:F6:6B:19:AE:DB:85:72:EB:DD:24:98:27:6A:B4:0E:AC:F5:2A:40 a=sendrecv ` ACK is the last message when the call finished.

[opera-capital]

And When I call with a Browser-Phone (2106) to SIP (2103):

Last INVITE:

`INVITE sip:2103@operacapital.com.br SIP/2.0 Via: SIP/2.0/WSS 192.0.2.75;branch=z9hG4bK6170248 To: sip:2103@operacapital.com.br From: "2106" sip:2106@operacapital.com.br;tag=mii5iiuund CSeq: 2 INVITE Call-ID: 4929qqc27r0qs6fsg7s1 Max-Forwards: 70 Authorization: Digest algorithm=MD5, username="2106", realm="asterisk", nonce="7713d5ab", uri="sip:2103@operacapital.com.br", response="f83f3a8c5b4b145ee90e89b1e21ab403" Contact: sip:6bah446i@192.0.2.75;transport=wss;ob Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER Supported: outbound User-Agent: Browser Phone 0.3.26 (SIPJS - 0.20.0) Mozilla/5.0 (Windows NT 10.0; Win64; x64) AppleWebKit/537.36 (KHTML, like Gecko) Chrome/120.0.0.0 Safari/537.36 Content-Type: application/sdp Content-Length: 2298

v=0 o=- 2750387442039835808 2 IN IP4 127.0.0.1 s=- t=0 0 a=group:BUNDLE 0 a=extmap-allow-mixed a=msid-semantic: WMS d5261fb1-def2-4161-9c5c-ccacfa478888 m=audio 38087 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 c=IN IP4 186.235.58.49 a=rtcp:9 IN IP4 0.0.0.0 a=candidate:648718536 1 udp 2122260223 192.168.4.100 65144 typ host generation 0 network-id 2 network-cost 50 a=candidate:3399025620 1 udp 2122194687 192.168.56.1 65145 typ host generation 0 network-id 3 a=candidate:3439334618 1 udp 2122129151 172.19.112.1 65146 typ host generation 0 network-id 4 a=candidate:2715923803 1 udp 2122063615 10.0.0.101 65147 typ host generation 0 network-id 1 network-cost 10 a=candidate:4124282988 1 udp 1685855999 186.235.58.49 38087 typ srflx raddr 10.0.0.101 rport 65147 generation 0 network-id 1 network-cost 10 a=candidate:1483033168 1 tcp 1518280447 192.168.4.100 9 typ host tcptype active generation 0 network-id 2 network-cost 50 a=candidate:3025592652 1 tcp 1518214911 192.168.56.1 9 typ host tcptype active generation 0 network-id 3 a=candidate:3016750658 1 tcp 1518149375 172.19.112.1 9 typ host tcptype active generation 0 network-id 4 a=candidate:3744356291 1 tcp 1518083839 10.0.0.101 9 typ host tcptype active generation 0 network-id 1 network-cost 10 a=ice-ufrag:3WMQ a=ice-pwd:K8p/w3o1iFNS43r6x3wi8sHK a=ice-options:trickle a=fingerprint:sha-256 17:6B:2E:FD:4A:CC:D4:76:D3:9F:0F:56:0B:A4:DE:AA:5D:6A:48:41:A1:96:72:47:66:53:B9:36:03:84:A0:58 a=setup:actpass a=mid:0 a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level a=extmap:2 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time a=extmap:3 http://www.ietf.org/id/draft-holmer-rmcat-transport-wide-cc-extensions-01 a=extmap:4 urn:ietf:params:rtp-hdrext:sdes:mid a=sendrecv a=msid:d5261fb1-def2-4161-9c5c-ccacfa478888 5a6ae723-68ae-479f-aa85-0c203e01f25a a=rtcp-mux a=rtpmap:111 opus/48000/2 a=rtcp-fb:111 transport-cc a=fmtp:111 minptime=10;useinbandfec=1 a=rtpmap:63 red/48000/2 a=fmtp:63 111/111 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:13 CN/8000 a=rtpmap:110 telephone-event/48000 a=rtpmap:126 telephone-event/8000 a=ssrc:3384071575 cname:NJWwjmMLGOLeDy23 a=ssrc:3384071575 msid:d5261fb1-def2-4161-9c5c-ccacfa478888 5a6ae723-68ae-479f-aa85-0c203e01f25a `

[opera-capital]

Hello

I found the problem! I tried to change the configuration by issabel http admin panel, but it doesn't change all the configuration.

So, I saw your video (https://www.youtube.com/watch?v=mS28vfT8wJ8&t=1217s) and change configurations manually.

When I changed manually the rtcp_mux to true and restart, the webrtc phone work!

Thanks!