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This is an automatically generated e-mail. To reply, visit:
<a href="https://reviewboard.asterisk.org/r/3447/">https://reviewboard.asterisk.org/r/3447/</a>
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<p style="margin-top: 0;">On April 16th, 2014, 6:02 p.m. CDT, <b>Joshua Colp</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3447/diff/2/?file=57476#file57476line11420" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/channels/chan_sip.c</a>
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(Diff revision 2)
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<td colspan="4"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">static int add_rpid(struct sip_request *req, struct sip_pvt *p)</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>if (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND)) {</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>/* If we trust, always use the true IP */</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>fromdomain = ast_sockaddr_stringify_host_remote(&p->ourip);</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>} else {</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));</pre></td>
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<td bgcolor="#ffffff" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span><span class="tb"> </span>fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "><span class="tb"> </span>}</pre></td>
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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;">If a legit fromdomain is specified you have to use it, or else you've just ignored the fromdomain configuration option.</pre>
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<p>On April 17th, 2014, 10:42 a.m. CDT, <b>Jonathan Rose</b> wrote:</p>
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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;">Hmmm. Since that's the case I need to figure out why I was seeing anonymous strings here with my test. This scares me a bit since I'm not sure how to deal with the anonymous.invalid address I was seeing in my baseline test for rpid (sendrpid=rpid, callingpres=prohib)
Remote-Party-ID: "123" <sip:123@anonymous.invalid>party=calling;privacy=full;screen=yes
If I need to send a non-anonymous address when using the trust_outbound_id setting, I need a way to get that, and right now it seems like one doesn't exist since the fromdomain will be invalid when callingpres is a prohibited value.</pre>
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<p>On April 17th, 2014, 10:56 a.m. CDT, <b>Jonathan Rose</b> wrote:</p>
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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;">You know, I suppose since anonymous.invalid and anonymous are at least predictable, I could just manually check to see if that's what is in fromdomain after doing
fromdomain = S_OR(p->fromdomain, ast_sockaddr_stringify_host_remote(&p->ourip));
and if that's the case when trust_id_outbound is set, then we reset to ast_sockaddr_stringify_host_remote... basically turn it into this:
fromdomain = p->fromdomain;
if (!fromdomain || (ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) && (strncmp(fromdomain, "anonymous", 9))) {
fromdomain = ast_sockaddr_stringify_host_remote(&p->ourip);
}
Not sure if that seems like an appropriate approach, but it's about all I've got for now.
I'm loathe to change the behavior of how things are working for sendrpid=rpid, but it seems apparent now that we would only use an anonymous domain name in my test for:
| pres=allowed | pres=prohibited |
----------------------+----------------------------------+-----------------------------------------------------------------------------------------+
trust_id_outbound=no | Remote-Party-ID: "123" <sip... | Remote-Party-ID: "123" <sip:123@anonymous.invalid>party=calling;privacy=full;screen=yes | <--- This one
----------------------+----------------------------------+-----------------------------------------------------------------------------------------+
trust_id_outbound=yes | Remote-Party-ID: "123" <sip... | Remote-Party-ID: "123" <sip:123@xxx.xxx.xxx.xxx>;party=calling;privacy=full;screen=yes |
----------------------+----------------------------------+-----------------------------------------------------------------------------------------+
is because the fromdomain was set to anonymous.invalid by some other mechanism. It might be the case that we actually need to use a deliberately anonymized address here, in which case we may need to bring back that anonymous string you were talking about in the finding below and make use of it here... in which case maybe we really expect a value of:
Remote-Party-ID: "Anonymous" <sip:anonymous@anonymous.invalid>;party=calling;privacy=full;screen=yes</pre>
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<pre style="margin-left: 1em; white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Oh, I missed wdoekes follow up post on this. It seems his suggested behavior is to not append the RPID header when prohib + distrust.
He also suggested maybe adding a legacy option so that we can keep sending the stuff we are already sending, which makes my head spin a little, but it might still be the best approach.</pre>
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<p>- Jonathan</p>
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<p>On April 16th, 2014, 4:23 p.m. CDT, Jonathan Rose wrote:</p>
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<div>Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark Michelson, and wdoekes.</div>
<div>By Jonathan Rose.</div>
<p style="color: grey;"><i>Updated April 16, 2014, 4:23 p.m.</i></p>
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<b style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Bugs: </b>
<a href="https://issues.asterisk.org/jira/browse/AST-1301">AST-1301</a>,
<a href="https://issues.asterisk.org/jira/browse/ASTERISK-19465">ASTERISK-19465</a>
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<b style="color: #575012; font-size: 10pt;">Repository: </b>
Asterisk
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Description </h1>
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<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;">Walter Doekes pointed out that this might cause a less than ideal situation in which people who were expecting P-Asserted-Identity not to disclose party information will now be sending privacy information, so I pulled this patch from 1.8-trunk and we will now review it here.
Without this patch, P-Asserted-Identity would always use anonymous for the caller ID information, and RFC-3325 seems to indicate that P-Asserted-Identity is something that should not be anonymized, but also only sent to trusted parties. The way this was presented to me, the intent here is that if you set callerpres to prohibited for a peer that receives P-Asserted-Identity, the P-Asserted-Identity shouldn't be anonymized, only the normal From/Contact headers would be anonymized. This apparently
The obvious method for dealing with this mid-release change is to make the change into an option which defaults off in 1.8-12 while defaulting on in trunk. Also I'll need to add Upgrade notes for trunk since this might not always be a desired behavior as well as CHANGES notes throughout to indicate the new option if that's what we settle on.</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Testing </h1>
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<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;">Call from SIP peer A to SIP peer B
settings for both peers:
sendrpid = pai
callerpres = prohib
Invite sent from Asterisk to the recipient of the call
------------------------------------------------------
Prior to patch:
Audio is at 19640
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.24.18.240:5060:
INVITE sip:123@10.24.18.240:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.18.246:5060;branch=z9hG4bK2fb42910;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as13075548
To: <sip:123@10.24.18.240:5060>
Contact: <sip:anonymous@10.24.18.246:5060>
Call-ID: 762b8a5e5848d7997f38f71a770d4dd9@10.24.18.246:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r410380
Date: Tue, 11 Mar 2014 22:59:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: "Anonymous" <sip:anonymous@anonymous.invalid>
Content-Type: application/sdp
Content-Length: 276
v=0
o=root 473543868 473543868 IN IP4 10.24.18.246
s=Asterisk PBX SVN-branch-1.8-r410380
c=IN IP4 10.24.18.246
t=0 0
m=audio 19640 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv
After patch:
Audio is at 11822
Adding codec 0x4 (ulaw) to SDP
Adding non-codec 0x1 (telephone-event) to SDP
Reliably Transmitting (NAT) to 10.24.18.240:5060:
INVITE sip:123@10.24.18.240:5060 SIP/2.0
Via: SIP/2.0/UDP 10.24.18.246:5060;branch=z9hG4bK5d4a7db8;rport
Max-Forwards: 70
From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=as181a14e3
To: <sip:123@10.24.18.240:5060>
Contact: <sip:anonymous@10.24.18.246:5060>
Call-ID: 721bef28208f7633288e929c6e88824e@10.24.18.246:5060
CSeq: 102 INVITE
User-Agent: Asterisk PBX SVN-branch-1.8-r410380M
Date: Tue, 11 Mar 2014 22:57:39 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH
Supported: replaces, timer
P-Asserted-Identity: "Goldy Locks" <sip:6018@10.24.18.246>
Privacy: id
Content-Type: application/sdp
Content-Length: 279
v=0
o=root 1606369071 1606369071 IN IP4 10.24.18.246
s=Asterisk PBX SVN-branch-1.8-r410380M
c=IN IP4 10.24.18.246
t=0 0
m=audio 11822 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
a=ptime:20
a=sendrecv</pre>
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<h1 style="color: #575012; font-size: 10pt; margin-top: 1.5em;">Diffs</b> </h1>
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<li>/branches/1.8/configs/sip.conf.sample <span style="color: grey">(412438)</span></li>
<li>/branches/1.8/channels/sip/include/sip.h <span style="color: grey">(412438)</span></li>
<li>/branches/1.8/channels/chan_sip.c <span style="color: grey">(412438)</span></li>
<li>/branches/1.8/CHANGES <span style="color: grey">(412438)</span></li>
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<p><a href="https://reviewboard.asterisk.org/r/3447/diff/" style="margin-left: 3em;">View Diff</a></p>
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