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

[Dmitriy Serov (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-27971?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=245124#comment-245124 ] 

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

res/res_pjsip_outbound_registration.c
static const char *fetch_google_access_token(const struct ast_sip_auth *auth)
char buf[4096];

4096 - sometimes goole returns more then 4096 bytes answer. I don't know why.
As the result:
ERROR[17668]: res_pjsip_outbound_registration.c:1462 in fetch_google_access_token: Could not retrieve Google OAuth 2.0 authentication

The same error in res_xmpp.c

Every asterisk update needs to patch: buffer size to 8192 bytes.

> 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
>         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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