[asterisk-bugs] [JIRA] (ASTERISK-18987) Alcatel workaround broke EARLY MEDIA for BeroFix

[Joshua Colp (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-18987?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Joshua Colp updated ASTERISK-18987:
-----------------------------------

    Affects Version/s: 13.18.4

> Alcatel workaround broke EARLY MEDIA for BeroFix
> ------------------------------------------------
>
>                 Key: ASTERISK-18987
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-18987
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/Interoperability
>    Affects Versions: 1.8.7.1, 10.0.0, 13.18.4
>         Environment: Asterisk connected to TELCO via Bero*Fix ver. 2.2rc3 (SIP<->ISDN30 gateway)
> Outgoing call (from Asterisk to Telco), destination number is unreachable number, announcement (EARLY MEDIA)
>            Reporter: Dan Lukes
>              Labels: EarlyMedia, asterisk, sdp
>   Original Estimate: 10m
>  Remaining Estimate: 10m
>
> Problem affects all versions from 1.4 to HEAD.
> During call setup (called number is unreachable which is announced by TELCO via EARLY MEDIA), the caller can hear short fragment (less than 0.5s) of early-media announcement only, then it is replaced by RINGING tone despite the announcement from TELCO continues.
> Schematic description of PRI handshaking (just for completeness):
> Bero*Fix <->Telco
> -> SETUP
> <- CALL PROCEEDING (B channel allocated) 
> <- PROGRESS (with Progress Indicator = 8 = EARLY MEDIA)
> ...
> <- DISCONNECT
> It is translated to following SIP handshaking:
> Asterisk <-> Bero*Fix:
> -> INVITE
> <- 100 Trying
> <- 183 PROCEEDING (with SDP)  
> <- 183 PROGRESS (no SDP as audio channel is opened already) 
> {color:blue}... about 15 second (it correlate with length of announcement){color}
> <- 404 RELEASE (hangup cause 31)
> The problem is caused by following fragment of code:
> {code:title=chan_sip.c, function handle_response_invite()|borderStyle=solid}
> } else
>   /* Alcatel PBXs are known to send 183s with no SDP after sending
>    * a 100 Trying response. We're just going to treat this sort of thing
>    * the same as we would treat a 180 Ringing
>    */
>   if (!req->ignore && p->owner) {
>      ast_queue_control(p->owner, AST_CONTROL_RINGING);
>   }
> {code}
> Proposed correction:
> 183 with no SDP should not trigger AST_CONTROL_RINGING when EARLY MEDIA active already
> so plain {color:red}"else"{color} above should be changed to:
> {color:green}
> } else if (p->invitestate != INV_EARLY_MEDIA)
> {color} 



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