[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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