[asterisk-bugs] [JIRA] (ASTERISK-27971) res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Wed Aug 28 11:57:53 CDT 2019
[ https://issues.asterisk.org/jira/browse/ASTERISK-27971?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team updated ASTERISK-27971:
-------------------------------------
Target Release Version/s: 17.0.0
> 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
> Target Release: 17.0.0
>
> Attachments: gvoice.txt
>
>
> 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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