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

Tue Apr 15 12:43:00 CDT 2014

> On April 15, 2014, 12:15 p.m., Mark Michelson wrote:
> > First off, I agree that anonymizing P-Asserted-Identity is not the correct way to be going here. The concept of trust is something that is not well-defined in chan_sip. The closest thing we currently have is the trustrpid option. When it is set, it means that we will actually use inbound Remote-Party-ID/P-Asserted-Identity headers as the identity of the entity that sent the SIP request/response to us.
> > 
> > In chan_pjsip, we have two trust-related options, trust_id_inbound and trust_id_outbound. trust_id_inbound is pretty much the same as chan_sip's trustrpid option. trust_id_outbound is a bit different. If set to yes, then we will send private identity information but include an indication that the information is private/restricted. In the case of P-Asserted-Identity, this means that the P-Asserted-Identity header is kept intact and a Privacy: id header is added. If trust_id_outbound is set to no, then we will send unrestricted identifying information, but anything marked as private/restricted does not get sent to the untrusted party at all.
> > 
> > Implementing an option similar to chan_pjsip's trust_id_outbound option is probably a good way to go. I think that it probably should default to off/false/no in all versions of Asterisk.

Alright, so if I'm interpreting you correctly here, what you are advocating is this:

* Add option trust_id_outbound (or similar name) to SIP peers
when on and sendrpid=pai, include P-Asserted-Identity with legitimate caller ID information as well as 'Privacy: id' header
when off and sendrpid=pai, don't bother with P-Asserted-Identity because the peer isn't a trusted party

Maybe I'm a little confused on that second part... What does it mean to send 'unrestricted identifying information' in the case when trust_id_outbound=no?

- Jonathan

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On April 15, 2014, 11:52 a.m., Jonathan Rose wrote:
> 
> -----------------------------------------------------------
> This is an automatically generated e-mail. To reply, visit:
> https://reviewboard.asterisk.org/r/3447/
> -----------------------------------------------------------
> 
> (Updated April 15, 2014, 11:52 a.m.)
> 
> 
> Review request for Asterisk Developers, Joshua Colp, Matt Jordan, Mark Michelson, and wdoekes.
> 
> 
> Bugs: AST-1301
>     https://issues.asterisk.org/jira/browse/AST-1301
> 
> 
> 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 412331 
>   /branches/1.8/channels/chan_sip.c 412331 
> 
> 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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