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List:       sr-users
Subject:    Re: [SR-Users] Resolved: JsSIP:UA Request-URI does not point to us
From:       Xuo Guoto <xuoguoto () protonmail ! com>
Date:       2022-03-24 16:16:39
Message-ID: aKTzO_A0whqr6YoV0_92eKtWu8Tfw8B1ahr7QHS1xdLzDNFUqfs60AKpBaObwLPn_XYPMQRPae6G00pX1ubNvPnYCmyTHFZzG91pEr1QEZk= () protonmail ! com
[Download RAW message or body]

Did some more digging into the source code of JsSIP. The condition that triggers the \
error message is: https://github.com/versatica/JsSIP/blob/3ab1fa7c8e09231c41ca21657bf962323906d5fe/lib/UA.js#L560
  /**
   * Request reception
   */
  receiveRequest(request)
  {
    const method = request.method;

    // Check that request URI points to us.
    if (request.ruri.user !== this._configuration.uri.user &&
        request.ruri.user !== this._contact.uri.user)
    {
      logger.debug('Request-URI does not point to us');
      if (request.method !== JsSIP_C.ACK)
      {
        request.reply_sl(404);
      }

      return;
    }

If I understand correctly if ruri.user is neither user in config or user in contact \
the request gets rejected.

R-URI is:
ACK sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 \
SIP/2.0

The SIP URI configured in tryjsip is \
sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com

This does not match, but if I change the SIP URI to
sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com (all small case) it works fine. \
I am not sure if this is a bug or violates the any RFC



Sent with ProtonMail secure email.

------- Original Message -------

On Thursday, March 24th, 2022 at 9:10 PM, Xuo Guoto <xuoguoto@protonmail.com> wrote:

> This is the ACK packet that is not getting recognized by JsSIP
> 
> ACK sip:stg-cqd0r2-10020005@erx-staging-q01.mydomain.com;gr=urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33 \
> SIP/2.0 
> Via: SIP/2.0/WSS 68.19.59.72:443;branch=z9hG4bKc3f.244f565f3d688006fb9c33138458f554.0
>  
> Via: SIP/2.0/UDP 127.0.0.8;branch=z9hG4bKsr-j4IPOlV7MGQKatycM.y7M.y7MmZfMxvwMxyAzweI \
> 36KYpEKqH.FAOBFAOBF7M.yXKFcQgSW6zweAowe4H.NAMxyX3heroEWvH9vsCFN43q1PCEjuMlWEWB0rO.aJgBc1WSPAMG4Z3RjLO.pqWSPlMRFwWEergc**
>  
> From: "User2" sip:9747815015@erx-staging-q01.mydomain.com;tag=27ec81cf-9ccc-45e6-b712-fe23337ed7d9
>  
> To: sip:Stg-CQD0r2-10020005@10.10.1.9;tag=6gc6gshfkb
> 
> Call-ID: 518500d1-9813-4af8-a249-981ca2ee8a4b
> 
> CSeq: 19353 ACK
> 
> Max-Forwards: 69
> 
> User-Agent: Asterisk PBX 18.8.0
> 
> Content-Length: 0
> 
> tryit-jssip.js:8 JsSIP:UA Request-URI does not point to us +40s
> 
> ------- Original Message -------
> 
> On Thursday, March 24th, 2022 at 8:20 PM, Xuo Guoto xuoguoto@protonmail.com wrote:
> 
> > Hi,
> > 
> > It seems when I paste the message in the web client, it got removed. Now trying \
> > again in text mode. 
> > REGISTER sip:erx-staging-q01.mydomain.com SIP/2.0
> > 
> > Via: SIP/2.0/WSS ol3dhprvu7jv.invalid;branch=z9hG4bK3674021
> > 
> > Max-Forwards: 69
> > 
> > To: sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com
> > 
> > From: "User" sip:Stg-CQD0r2-10020005@erx-staging-q01.mydomain.com;tag=65u34oje2s
> > 
> > Call-ID: b624vmbvuioma46354gmi5
> > 
> > CSeq: 1 REGISTER
> > 
> > Contact: sip:93he4k0p@ol3dhprvu7jv.invalid;transport=ws;+sip.ice;reg-id=1;+sip.instance="urn:uuid:b8a691ff-f678-4866-8cf0-780d670e7e33";expires=600
> >  
> > Expires: 600
> > 
> > Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO,NOTIFY
> > 
> > Supported: path,gruu,outbound
> > 
> > User-Agent: JsSIP 3.9.0
> > 
> > Content-Length: 0
> > 
> > I hadn't noticed that some text was removed by the client.
> > 
> > X.
> > 
> > ------- Original Message -------
> > 
> > On Thursday, March 24th, 2022 at 6:31 PM, Daniel-Constantin Mierla \
> > miconda@gmail.com wrote: 
> > > Hello,
> > > 
> > > is the REGISTER without a Contact URI or the message you pasted omitted it?
> > > 
> > > Cheers,
> > > 
> > > Daniel
> > > 
> > > On 22.03.22 10:29, Xuo Guoto wrote:
> > > 
> > > > Hello all,
> > > > 
> > > > I am facing an issue with JsSIP not recognizing replies from Kamailio. the \
> > > > call sequence goes as follows: 
> > > > INVITE -----------------------------><-------------------------------SIP/2.0 \
> > > > 100 Trying<-------------------------------SIP/2.0 180 \
> > > > Ringing<-------------------------------SIP/2.0 200 OKACK \
> > > > --------------------------------><-------------------------------SIP/2.0 200 \
> > > > OKACK --------------------------------><-------------------------------SIP/2.0 \
> > > > 200 OKACK --------------------------------><-------------------------------SIP/2.0 \
> > > > 200 OKACK --------------------------------><-------------------------------SIP/2.0 \
> > > > 200 OKACK --------------------------------><-------------------------------SIP/2.0 \
> > > > 200 OKACK --------------------------------><-------------------------------SIP/2.0 \
> > > > 200 OKACK --------------------------------><-------------------------------BYE404 \
> > > > Not Found ----------------------> 
> > > > When JsSIP receives ACK it prints an error: JsSIP:UA Request-URI does not \
> > > > point to us

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