VOIP Server Registration

[massimiliano-andreatta]

Hi,
I wanted to let you know that I'm experiencing some issues with your code when registering to a VOIP server. In order to get the system to register correctly, I had to modify a part of the code inside SIP -> Builder, adding the following code:

DigestResponseShort: (username, password, realm, nonce, method, uri) => { const ha1 = crypto.createHash("md5").update(${username}:${realm}:${password}).digest("hex"); const ha2 = crypto.createHash("md5").update(${method}:${uri}).digest("hex"); const response = crypto.createHash("md5").update(${ha1}:${nonce}:${ha2}).digest("hex"); return response; }

Then, I used it inside SIPMessageObject:

if (challenge_data) { var cnonce = Builder.generateBranch(); if (proxy_auth) { const response = Builder.DigestResponse( props.username, props.password, challenge_data.realm, challenge_data.nonce, res.method, res.requestUri, challenge_data.qop, cnonce, '00000001' ); res.headers['Proxy-Authorization'] = Digest username="${props.username}", realm="${challenge_data.realm}", nonce="${challenge_data.nonce}", uri="${res.requestUri}", response="${response}", qop="${challenge_data.qop}", cnonce="${cnonce}", nc="00000001", algorithm="MD5"; } else { const response = Builder.DigestResponseShort( props.username, props.password, challenge_data.realm, challenge_data.nonce, "REGISTER", ${res.requestUri}); res.headers['Authorization'] =Digest username="${props.username}", realm="${challenge_data.realm}", nonce="${challenge_data.nonce}", uri="${res.requestUri}", response="${response}", algorithm="MD5"; } }

[Nik-Hendricks]

What is the vendor of the SIP server you are trying to register with. I have not encountered any issues with registration to my grandstream PBX. or one of my SIP trunk using proxy authentication.

If this is an edge case for a certain vendor. I would like to know so we can add it in. I would love to accept a contribution from you with clear code if this is in fact an edge case.

[massimiliano-andreatta]

My VOIP trunk provider is "sip.vivavox.it." What do you need to verify this?
Do you want the Wireshark packets?

[Nik-Hendricks]

Don't post it here unless you change your register password on the sip trunk end. change it to something fake that you can change back later just to make sure you don't leak anything private. Also, If you have a version that works with both schemes and clean code, I would accept your pull request and you would be a contributor.

[massimiliano-andreatta]

The domain I provided is from an Italian SIP provider, and I won't give you the username/password. If you want me to contribute, that's no problem. Right now, I'm trying to answer a call, but even though I send the "Trying" and "Ringing" packets, it keeps going into a loop.

[massimiliano-andreatta]

I managed to make the phone ring. Now let's move on to answering the call and then capturing the audio.

[Nik-Hendricks]

You should add me on Discord nightcrawler43 in the /RTP folder. it exports the SessionManager and Session class. These are what we are using for for the RTP session as far as streaming and receiving. You could use 'ffplay' to play the RTP stream on your custom SIP client. i plan to add that and a RTP proxy feature to the RTP section of the code soon as well. That way you could proxy the audio from one device to the other for easier debugging in case of a remote client that you cant directly listen on.