[asterisk-bugs] [JIRA] (ASTERISK-27971) res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability

Dmitriy Serov (JIRA) noreply at issues.asterisk.org
Fri Aug 24 11:58:54 CDT 2018


    [ https://issues.asterisk.org/jira/browse/ASTERISK-27971?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=244598#comment-244598 ] 

Dmitriy Serov commented on ASTERISK-27971:
------------------------------------------

After applying patch and using new "outbound_registration" for all endpoints where is the BUG.

INVITE sip:791519XXXX at multifon.ru SIP/2.0
Supported: 100rel, timer, replaces, norefersub
Session-Expires: 1800
Min-SE: 90
Route: <sip:srv_d24948 at 193.201.XX.XX:8080;transport=udp;lr>;line=zxzfbbm
P-Preferred-Identity: 
Max-Forwards: 70
User-Agent: ruVoIP.net PBX
Content-Type: application/sdp
Content-Length:   245

Header "P-Preferred-Identity" has empty value.
As the result: SIP/2.0 400 Invalid P-Preferred-Identity
It is not google voice account.

> res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
> -----------------------------------------------------------------------------
>
>                 Key: ASTERISK-27971
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27971
>             Project: Asterisk
>          Issue Type: New Feature
>      Security Level: None
>          Components: pjproject/pjsip, Resources/res_pjsip_outbound_registration
>    Affects Versions: GIT
>            Reporter: Nick French
>            Assignee: Nick French
>              Labels: pjsip
>
> Background: Google Voice trunks are currently supported in Asterisk via chan_motif. Google has announced they plan to migrate away from the XMPP protocol used by chan_motif to a new SIP-based protocol. However, their new SIP servers use additional standards extending what is commonly implemented in a SIP UAC.
> The following additional features required by the new Google Voice SIP registrar are not currently implemented in Asterisk:
> - Service-Routes (RFC 3608)
> - P-Preferred-Identity (RFC 3325)
> - Outbound supported header (RFC 5626)
> - OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02)
> - Mechanisms to use separate TLS transports for separate registrations and their associated message dialog
> - (optional) User-configurable additions to SIP Contact header



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