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List: asterisk-dev
Subject: Re: [asterisk-dev] [Code Review] 2827: chan_sip: Reject call on 200 OK response to invite that lacks
From: "Olle E Johansson" <reviewboard () asterisk ! org>
Date: 2013-09-06 6:07:08
Message-ID: 20130906060708.4510.36537 () hotblack ! digium ! com
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This doesn't work well with PRACK or UPDATES. Asterisk could already have gotten an \
SDP in another message. (I have a branch with PRACK support that is in production). \
We need to check if we already have gotten an SDP before we decide to hang up the \
call. I also need to check the RFCs for this situation.
- Olle E Johansson
On Sept. 5, 2013, 9:31 p.m., jrose wrote:
>
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/2827/
> -----------------------------------------------------------
>
> (Updated Sept. 5, 2013, 9:31 p.m.)
>
>
> Review request for Asterisk Developers, Joshua Colp, Matt Jordan, and Mark \
> Michelson.
>
> Bugs: ASTERISK-22424
> https://issues.asterisk.org/jira/browse/ASTERISK-22424
>
>
> Repository: Asterisk
>
>
> Description
> -------
>
> One of our SIP tests was previously pushing 200 OKs without SDP and Asterisk would \
> accept these calls without question. According to Mark this should not be accepted \
> because there will be no way to know where to send media to or receive media from \
> in these circumstances. The approach this patch takes is to forcibly hang up the \
> call at this point if there is no SDP on the response provided that it's not a \
> response to a reinvite (in which case the behavior is the same as if there were an \
> SDP that couldn't be parsed properly).
>
> Diffs
> -----
>
> /branches/1.8/channels/chan_sip.c 398378
>
> Diff: https://reviewboard.asterisk.org/r/2827/diff/
>
>
> Testing
> -------
>
> Tested it against SIP_hold before and after
> Tested it against a number of testsuite tests against SIP (any of the ones I could \
> run before the patch) Tested regular SIP phone calls (they didn't hit the modified \
> code path though).
>
> Thanks,
>
> jrose
>
>
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This is an automatically generated e-mail. To reply, visit:
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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: \
-pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">This doesn't work \
well with PRACK or UPDATES. Asterisk could already have gotten an SDP in another \
message. (I have a branch with PRACK support that is in production). We need to check \
if we already have gotten an SDP before we decide to hang up the call. I also need to \
check the RFCs for this situation.</pre> <br />
<p>- Olle E</p>
<br />
<p>On September 5th, 2013, 9:31 p.m. CEST, jrose wrote:</p>
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<div>Review request for Asterisk Developers, Joshua Colp, Matt Jordan, and Mark \
Michelson.</div> <div>By jrose.</div>
<p style="color: grey;"><i>Updated Sept. 5, 2013, 9:31 p.m.</i></p>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-22424">ASTERISK-22424</a>
</div>
<div style="margin-top: 1.5em;">
<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
</div>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" \
style="border: 1px solid #b8b5a0"> <tr>
<td>
<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: \
-moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: \
break-word;">One of our SIP tests was previously pushing 200 OKs without SDP and \
Asterisk would accept these calls without question. According to Mark this should not \
be accepted because there will be no way to know where to send media to or receive \
media from in these circumstances. The approach this patch takes is to forcibly hang \
up the call at this point if there is no SDP on the response provided that it's \
not a response to a reinvite (in which case the behavior is the same as if there were \
an SDP that couldn't be parsed properly).</pre> </td>
</tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
<table width="100%" bgcolor="#ffffff" cellspacing="0" cellpadding="10" style="border: \
1px solid #b8b5a0"> <tr>
<td>
<pre style="margin: 0; padding: 0; white-space: pre-wrap; white-space: \
-moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: \
break-word;">Tested it against SIP_hold before and after Tested it against a number \
of testsuite tests against SIP (any of the ones I could run before the patch) Tested \
regular SIP phone calls (they didn't hit the modified code path though).</pre> \
</td> </tr>
</table>
<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
<ul style="margin-left: 3em; padding-left: 0;">
<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(398378)</span></li>
</ul>
<p><a href="https://reviewboard.asterisk.org/r/2827/diff/" style="margin-left: \
3em;">View Diff</a></p>
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