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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 18th, 2014, 2:06 a.m. CDT, <b>wdoekes</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;">Looks mostly good. The documentation could use some tweaking.
And I'm unsure what you did with fromdomain there.
Also, unrelated, your latest chart said this:
> <pps:public> == party=calling;privacy=off;screen=no
> <pps:private> == party=calling;privacy=full;screen=yes
While your comment said this:
> Well, I had no plans of making any changes to how
> party/privacy/screen are set, so he can rest easy on that one.
That did still leave the possibility for extra confusion in this regard.
'screen=whatever' or 'screen=no' in both places would have been more
appropriate. But yes, Pavel can rest easy.</pre>
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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;">Yes, sorry, that was more for my reference when writing tests than anything else. None of those values are in any way touched by the patch.</pre>
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<p style="margin-top: 0;">On April 18th, 2014, 2:06 a.m. CDT, <b>wdoekes</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3447/diff/3/?file=57618#file57618line15" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/CHANGES</a>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">SIP Changes</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">-----------</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> fields for prohibited callingpres information. If a peer is set to be</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> trusted with this setting, then caller ID information in Remote-Party-ID</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> and P-Asserted-Identity fields will not be anonymized to the recipient. In</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> the case of P-Asserted-Identity, a Privacy: id header will be appended as</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> well. If a peer is set not to be trusted, then PAI/RPID are omitted when</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> the remote party has prohibited callingpres information. By default,</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> legacy mode is used which leaves the behavior unchanged from 1.8.26.1</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;">> By default, legacy is used.
> trust_id_outbound=legacy: behavior
> remains the same as 1.8.26.1 - When
> dealing with prohibited callingpres,
> RPID/PAI headers are created for both
> sendrpid=pai and sendrpid=rpid are
> appended, but the data is anonymized.
sendrpid=rpid + legacy is *not* anonymous, while sendrpid=pai + legacy *is*.
Clarity in here isn't as important to me as clarity in sip.conf though.</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;">sendrpid=rpid + legacy is slightly anonymized... just not fully anonymized. When the peer's callingpres is set to one of the private types like 'prohib', then the fromdomain is set to FROMDOMAIN_INVALID (see initreqprep) which is anonymous.invalid. I'm not sure if there are any conditions where fromdomain will be changed again... in which case legacy wouldn't anonymize it at that point, but the presence of anonymous.invalid is why there was a need for the stuff I described in the finding below.</pre>
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<p style="margin-top: 0;">On April 18th, 2014, 2:06 a.m. CDT, <b>wdoekes</b> wrote:</p>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; ">SIP Changes</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> * SIP peers can now specify 'trust_id_outbound' which affects RPID/PAI</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;">Should this be "chan_sip" changes? Since there is that other driver in asterisk 12.</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;">Fair point. There were previous logs for chan_sip called just 'SIP Changes', but you are right, precedent goes out the window in this case.</pre>
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<p style="margin-top: 0;">On April 18th, 2014, 2:06 a.m. CDT, <b>wdoekes</b> wrote:</p>
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<a href="https://reviewboard.asterisk.org/r/3447/diff/3/?file=57619#file57619line11428" style="color: black; font-weight: bold; text-decoration: underline;">/branches/1.8/channels/chan_sip.c</a>
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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><span class="tb"> </span><span class="tb"> </span>((ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_YES) &&</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><span class="tb"> </span>!strcmp("anonymous.invalid", fromdomain))) {</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 the fromdomain is NULL or if it was set to anonymous.invalid due to privacy settings and we trust the peer,</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> * use the host IP address */</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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<pre style="white-space: pre-wrap; white-space: -moz-pre-wrap; white-space: -pre-wrap; white-space: -o-pre-wrap; word-wrap: break-word;">Aren't you altering legacy here?
Previously we would always use host_remote if p->fromdomain was empty. Now we only use it when trust_id_outbound=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;">No, that condition starts in the line above, so it might be the case that you just missed the first part of the condition.
if (!fromdomain ||
((ast_test_flag(&p->flags[1], SIP_PAGE2_TRUST_ID_OUTBOUND) == SIP_PAGE2_TRUST_ID_OUTBOUND_YES) &&
!strcmp("anonymous.invalid", fromdomain))) {
/* If the fromdomain is NULL or if it was set to anonymous.invalid due to privacy settings and we trust the peer,
* use the host IP address */
fromdomain = ast_sockaddr_stringify_host_remote(&p->ourip);
}
(!fromdomain || (...)) is true in the case you are describing (fromdomain = p->fromdomain which is empty), so we use host_remote just as before.</pre>
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<p style="margin-top: 0;">On April 18th, 2014, 2:06 a.m. CDT, <b>wdoekes</b> wrote:</p>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> ; legacy - RPID/PAI will be included for private peer information, but</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> ; private data that would be included in them will be anonymized. This</pre></td>
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<td bgcolor="#c5ffc4" width="50%"><pre style="font-size: 8pt; line-height: 140%; margin: 0; "> ; may violate RFC-3325, but this follows historic behavior.</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;">in case of PAI, private data will be anonymized (following historic behaviour violating RFC-3325)
in case of RPID, this behaves as trust_id_outbound=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;">In the case of RPID, not entirely. If trust_id_outbound is legacy and the fromdomain is anonymous.invalid coming into the function that appends the RPID header, the domain chosen will be anonymous.invalid. If trust_id_outbound=yes, then because fromdomain is anonymous.invalid, the host address will be used instead.
I found it all a little confusing myself, but it was the best I could make of a bad situation.</pre>
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<p>- Jonathan</p>
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<p>On April 17th, 2014, 3:25 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 17, 2014, 3:25 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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