[asterisk-bugs] [JIRA] (ASTERISK-20644) Don't always use the existing TCP connection for in-dialog requests

[Iñaki Baz Castillo (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-20644?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=199170#comment-199170 ] 

Iñaki Baz Castillo commented on ASTERISK-20644:
-----------------------------------------------

IMHO related "wrong" code is here:

http://lists.digium.com/pipermail/asterisk-commits/2012-October/057327.html

specially here:

{code}
if (p->socket.type != SIP_TRANSPORT_UDP && p->socket.tcptls_session) {
      /* For TCP/TLS sockets that are connected we won't need
       * to do any hostname/IP lookups */
{code}

This is incorrect. For sending in-dialog request Asterisk should not reuse the remote client/proxy/server initiated connection but open a new one against the URI in the top Route or against the URI in the Contact if there is no Route header. Obviously this can be overriden in case of "nat=something..." but it should NOT be the default and only behavior.
                
> Don't always use the existing TCP connection for in-dialog requests
> -------------------------------------------------------------------
>
>                 Key: ASTERISK-20644
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20644
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 11.0.0
>            Reporter: Iñaki Baz Castillo
>
> If Asterisk receives an INVITE via TCP comming from a SIP proxy, answers the call and later Asterisk sends a re-INVITE or BYE for that dialog, Asterisk sends such an in-dialog request over the existing TCP connection previously opened by the SIP proxy. This is incorrect. Asterisk should open a NEW TCP connection to the address given in the top Record-Route header.
> So for example, Asterisk receives the following INVITE from TCP 1.2.3.4 port 8888:
> {code}
> # TCP 1.2.3.4:8888 => ASTERISK:5060
> INVITE sip:test at domain.com SIP/2.0
> Record-Route: <sip:1.2.3.4:5060;transport=tcp>
> {code}
> When Asterisk sends a BYE or a re-INVITE in this leg, it MUST respect the address in the top Route of the BYE or re-INVITE, which is: TCP 1.2.3.4 port 5060. So it should send the BYE to:
> {code}
> # TCP ASTERISK:5060 => 1.2.3.4:5060
> BYE sip:abc at 9.8.7.6 SIP/2.0
> Route: <sip:1.2.3.4:5060;transport=tcp>
> {code}
> This is something really basic in RFC 3261 and mandatory. Asterisk could use the existing TCP connection just in case nat=yes (or some other new values I don't fully know in the latest version), but by default, if "nat" parameter is not set for the Proxy peer, please DON'T reuse the existing TCP connection because that is a violation of RFC 3261.
> In fact, it's perfectly legal for a SIP proxy or SIP server to refuse/reject SIP requests coming from a TCP connection that the proxy/server itself opened against a remote proxy/server. These are RFC 3261 rules, really.

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