[asterisk-dev] [Code Review] 3447: Send real CallerID information with P-Asserted-Identity (RFC-3325)

Thu Apr 17 10:42:04 CDT 2014

> On April 16, 2014, 6:02 p.m., Joshua Colp wrote:
> > /branches/1.8/channels/chan_sip.c, lines 11420-11425
> > <https://reviewboard.asterisk.org/r/3447/diff/2/?file=57476#file57476line11420>
> >
> >     If a legit fromdomain is specified you have to use it, or else you've just ignored the fromdomain configuration option.

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 at 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.

> On April 16, 2014, 6:02 p.m., Joshua Colp wrote:
> > /branches/1.8/channels/chan_sip.c, lines 11430-11438
> > <https://reviewboard.asterisk.org/r/3447/diff/2/?file=57476#file57476line11430>
> >
> >     Previously if the presentation restriction was such that it wasn't allowed... then the anonymous header would get added. Reading this logic I don't see how the header would get added.

The anonymous string would only get used under the following conditions:

1. sendrpid mode was pai
2. the lid_pres value was not an 'AST_PRES_ALLOWED' class of callingpres values.

Going by the chart for pai that wdoekes suggested:

                      | pres=allowed            | pres=prohibited       |
----------------------+-------------------------+-----------------------+
trust_id_outbound=no  | PAI: 123, Privacy: none |                       | <-------------- It's this one
----------------------+-------------------------+-----------------------+
trust_id_outbound=yes | PAI: 123, Privacy: none | PAI: 123, Privacy: id |
----------------------+-------------------------+-----------------------+

Then we shouldn't be sending either header in this case, which is why the anonymous string was removed.

- Jonathan

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On April 16, 2014, 4:23 p.m., Jonathan Rose wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3447/
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> 
> (Updated April 16, 2014, 4:23 p.m.)
> 
> 
> Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark Michelson, and wdoekes.
> 
> 
> Bugs: AST-1301 and ASTERISK-19465
>     https://issues.asterisk.org/jira/browse/AST-1301
>     https://issues.asterisk.org/jira/browse/ASTERISK-19465
> 
> 
> Repository: Asterisk
> 
> 
> Description
> -------
> 
> 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.
> 
> 
> Diffs
> -----
> 
>   /branches/1.8/configs/sip.conf.sample 412438 
>   /branches/1.8/channels/sip/include/sip.h 412438 
>   /branches/1.8/channels/chan_sip.c 412438 
>   /branches/1.8/CHANGES 412438 
> 
> Diff: https://reviewboard.asterisk.org/r/3447/diff/
> 
> 
> Testing
> -------
> 
> 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 at 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 at anonymous.invalid>;tag=as13075548
> To: <sip:123 at 10.24.18.240:5060>
> Contact: <sip:anonymous at 10.24.18.246:5060>
> Call-ID: 762b8a5e5848d7997f38f71a770d4dd9 at 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 at 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 at 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 at anonymous.invalid>;tag=as181a14e3
> To: <sip:123 at 10.24.18.240:5060>
> Contact: <sip:anonymous at 10.24.18.246:5060>
> Call-ID: 721bef28208f7633288e929c6e88824e at 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 at 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
> 
> 
> Thanks,
> 
> Jonathan Rose
> 
>

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