[asterisk-bugs] [JIRA] (ASTERISK-27971) res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability
Dmitriy Serov (JIRA)
noreply at issues.asterisk.org
Wed Oct 10 14:13:54 CDT 2018
[ 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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