[asterisk-bugs] [JIRA] (ASTERISK-30042) Registration over websocket returns a rewritten contact

[Thomas Guebels (JIRA)]

Thomas Guebels created ASTERISK-30042:
-----------------------------------------

             Summary: Registration over websocket returns a rewritten contact
                 Key: ASTERISK-30042
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30042
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_pjsip_registrar, Resources/res_pjsip_transport_websocket
    Affects Versions: 18.7.1
            Reporter: Thomas Guebels


When registering over WebSocket using SIP.js, res_pjsip_transport_websocket rewrites the {{random.invalid}} contact that is sent in the REGISTER request. The original {{random.invalid}} host is replaced by the apparent ip:port from where the requests came from. This is then stored as the contact uri.

{code}
REGISTER sip:dev.elan.escaux.com SIP/2.0
Via: SIP/2.0/WSS k2km30gagse1.invalid;branch=z9hG4bK6030502
To: <sip:identity at localhost>
From: <sip:identity at localhost>;tag=31mhoq79ir
CSeq: 3 REGISTER
Call-ID: o5n3ln3aog19bdpt5bv6
Contact: <sip:bhe74v8j at k2km30gagse1.invalid;transport=ws>;expires=600
User-Agent: SIP.js/0.20.0
{code}

In the REGISTER response, this rewritten contact is sent back to SIP.js, triggering the following error, as SIP.js can't find back the contact it tried to register.

{code}
SIP/2.0 200 OK
Via: SIP/2.0/WSS k2km30gagse1.invalid;rport=42956;received=172.16.25.12;branch=z9hG4bK4399238
To: <sip:identity at localhost>;tag=z9hG4bK4399238
From: <sip:identity at localhost>;tag=31mhoq79ir
Call-ID: o5n3ln3aog19bdpt5bv6
CSeq: 3 REGISTER
Contact: <sip:bhe74v8j at 172.16.123.123:5247;transport=ws>;expires=599
Server: Asterisk PBX 18.7.1
{code}

bq. No Contact header pointing to us, dropping response
from https://github.com/onsip/SIP.js/blob/5555a9811f33df1f1461c9094a59b82307b25405/src/api/registerer.ts#L421

It is worth noting that this doesn't happen with the default SIP.js config in which it is more lenient and only verifies the user part of the contact. However, when you specify a custom contact user in the its configuration, it does the full check user at ip:port as per the RFC. This probably explains why one doesn't encounter that bug right away when trying SIP.js + asterisk.

I tested a fix which is to save the original contact host in an x-ast-orig-host contact parameter, pretty much like it is done in res_pjsip_nat. It fixes my problem. If you think it is the right way to solve this I can provide a patch.



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