Register authorization error

[Bodonyi Titusz]

I've followed your guide and I faced this issue regarding to registration.
It seems I was wrong with configuration to register, but I couldn't find any solution about authorization of registration.
I got this log:

Sun Jan 10 2021 08:18:26 GMT-0800 (Pacific Standard Time) | sip.Transport | Sending WebSocket message:

REGISTER sip:sip...com SIP/2.0

Via: SIP/2.0/WSS mf4kmiau48r5.invalid;branch=z9hG4bK9728582

To: <sip:9...@sip...com>

From: "John" < sip:9...@sip...com  >;tag=qcalgoeqed

CSeq: 3 REGISTER

Call-ID: 0ihmna2rf8rngnk944nk

Max-Forwards: 70

Authorization: Digest algorithm=MD5, username="903", realm="sip...com", nonce="3d120af1-a126-4ef1-9977-8c31d5f4c6ff", uri="sip:sip...com", response="10ad4cd728953a827ef5f64938e59d57", qop=auth, cnonce="5st0sn2viec0", nc=00000001

Contact: <sip:j4ea...@mf4kmiau48r5.invalid;transport=ws>;expires=600

Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

Supported: outbound, path, gruu

User-Agent: SIP.js/0.17.1

Content-Length: 0

logger-factory.js:80 Sun Jan 10 2021 08:18:26 GMT-0800 (Pacific Standard Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 401 Unauthorized

Via: SIP/2.0/WSS 16n1ocot4dm3.invalid;branch=z9hG4bK3354066;received=50.97.232.186;rport=25026

From: "John" < sip:9...@sip...com  >;tag=qml54hiivf

To: < sip:9...@sip...com  >;tag=gvg5DSQKr6Z7Q

Call-ID: s937kr2k8ujmrkt3fnde

CSeq: 2 REGISTER

User-Agent: FreeSWITCH

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

WWW-Authenticate: Digest realm="sip...com", nonce="cf038b1e-bb3b-4179-b860-b6aa5f051cc5", algorithm=MD5, qop="auth"

Content-Length: 0

logger-factory.js:80 Sun Jan 10 2021 08:18:26 GMT-0800 (Pacific Standard Time) | sip.Transport | Sending WebSocket message:

REGISTER sip:sip...com SIP/2.0

Via: SIP/2.0/WSS 16n1ocot4dm3.invalid;branch=z9hG4bK6167732

To: < sip:9...@sip...com  >

From: "John" < sip:9...@sip...com  >;tag=qml54hiivf

CSeq: 3 REGISTER

Call-ID: s937kr2k8ujmrkt3fnde

Max-Forwards: 70

Authorization: Digest algorithm=MD5, username="903", realm="sip...com", nonce="cf038b1e-bb3b-4179-b860-b6aa5f051cc5", uri="sip:sip...com", response="50e5f710d73518d0a9fafd93821fff08", qop=auth, cnonce="u4ok47dnd2n7", nc=00000001

Contact: <sip:5ksf...@16n1ocot4dm3.invalid;transport=ws>;expires=600

Allow: ACK,CANCEL,INVITE,MESSAGE,BYE,OPTIONS,INFO,NOTIFY,REFER

Supported: outbound, path, gruu

User-Agent: SIP.js/0.17.1

Content-Length: 0

logger-factory.js:80 Sun Jan 10 2021 08:18:27 GMT-0800 (Pacific Standard Time) | sip.Transport | Received WebSocket text message:

SIP/2.0 403 Forbidden

Via: SIP/2.0/WSS mf4kmiau48r5.invalid;branch=z9hG4bK9728582;received=50.97.232.186;rport=24974

From: "John" < sip:9...@sip...com  >;tag=qcalgoeqed

To: < sip:9...@sip...com  >;tag=H59XFm8pNFptK

Call-ID: 0ihmna2rf8rngnk944nk

CSeq: 3 REGISTER

User-Agent: FreeSWITCH

Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE

Supported: timer, path, replaces

Content-Length: 0

logger-factory.js:75 Sun Jan 10 2021 08:18:27 GMT-0800 (Pacific Standard Time) | sip.Registerer | Failed to register, status code 403

As you seen from the log, I used FreeSWITCH server and my attemption to connect to the server was success.

It would be great if you suggest any way to get rid of this ridiculous configuration issue, or give me any tip.

Thank you a lot in advance.

[Slavik Bialik]

Are you sure it is not misconfigurations on FreeSWITCH side? Try to go into your FreeSWITCH and do "fs_cli".

Then make an additional REGISTER attempt and see if you get some logs about it (you'll get probably a log colored in purple if I am right).

Anyway, the Authorization header looks fine. The username and the realm (unless the domain in the realm is not the domain configured for the directory users on your FreeSWITCH).

Can you also try and register to this user with a MicroSIP / Zoiper and see if it works?

[Bodonyi Titusz]

Hello Slavik 

Thank you for your kind response.

Yes I am sure about configurations of FreeSWITCH. I have tried voip phone with same credentials and that's why I am gonna fix something on client-side.

Here you can see my code in order to integrate SIP.js

        const target = "sip:903@sip...com";
        const webSocketServer = "wss://myWebsocketAdress";
        const displayName = "John";
        const password = "userPassword";
       
        const simpleUserOptions = {
          aor: target,
          delegate: {
            onCallCreated() {},
            onCallReceived() {
              ...
            },
            onCallAnswered() {
              ...
            },
            onCallHangup() {
              ...
            },
            onCallHold(held) {},
          },
          media: {
            remote: {
              audio: document.getElementById('remoteAudio'),
            },
          },
          userAgentOptions: {
            authorizationPassword: password,
            authorizationUsername: displayName
          },
        };

        const simpleUser = new Web.SimpleUser(
          webSocketServer,
          simpleUserOptions
        );

        simpleUser
          .connect()
          .catch(error => {
            console.error(`failed to connect`);
            console.error(error);
          })
          .then(() => {
            this.simpleUser.register().then(() => {
              simpleUser.delegate = {
                onCallReceived() {
                  ...
                },
                onCallAnswered() {
                  ...
                },
                onCallHangup() {
                  ...
                },
              };
            })
            .catch((e) => {
              console.warn(`failed to register`);
            });
         
I will be pleasure you let me know what's wrong with my code.

Thank you in advance.

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