[asterisk-dev] [Code Review] P-Asserted-Identity Privacy - fixed behaviour

[jamicque]

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This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/1803/
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(Updated March 7, 2012, 6:18 p.m.)


Review request for Asterisk Developers.


Changes
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All of Marks remarks have been fixed.
I've added trustpeer configuration variable.
Comments have been updated also.


Summary
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It seams that in Asterisk privacy with PAI is not implemented correctly.

According to RFC 3325 when using privacy, FROM header should be set to anonymous at anonymous.invalid and PAI header should be set to caller num and name. The privacy is implemented by adding privacy: id header.
Now when we use pai and callpres=prohib in P-Asserted-Identity header we have something which is not correct to any rfc.
P-Asserted-Identity: "Anonymous" <sip:anonymous at anonymous.invalid>

What my patch does:
1) it adds Privacy header when PAI is used (values "none" or "id" depending on callpres)
2)
3) "sendrpid" configuration option have been expanded:
now it can have those values:

    no - nothing changed
    yes - rpid header is added, when call PRES=prohi, FROM header is not changed
    rpid - the same as yes
    pai - pai header is added, when call PRES=prohi, FROM header is not changed

NEW VALUES:

    rpid,trusted (NEW) - the same as yes
    rpid,untrusted (NEW) - rpid header is added, when call PRES=prohi, FROM header is changed to anonymous at anonymous.invalid and rpid header is srtiped.
    pai,trusted (NEW) - the same as pai
    pai,untrusted (NEW) - pai header is added, when call PRES=prohi, FROM header is chenged to anonymous at anonymous.invalid and pai header is srtiped. - as in RFC 3325

When we are using PAI or RPID ,fromname is defined and CLIR, we do not set anonymous at anonymous.invalid - coz this from in this situation is usually used for authentication.


This addresses bug ASTERISK-19465.
    https://issues.asterisk.org/jira/browse/ASTERISK-19465


Diffs (updated)
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  /trunk/channels/sip/include/sip.h 358575 
  /trunk/channels/chan_sip.c 358575 
  /trunk/configs/sip.conf.sample 358608 

Diff: https://reviewboard.asterisk.org/r/1803/diff


Testing
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I've done some basing test with outgoing calls and everything seems to wroks fine.


Thanks,

jamicque

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