[asterisk-bugs] [Asterisk 0018228]: SIP 180 Sent from provider, Asterisk sends 180 and 183 w/SDP. No ringback on calls

[Asterisk Bug Tracker]

A NOTE has been added to this issue. 
====================================================================== 
https://issues.asterisk.org/view.php?id=18228 
====================================================================== 
Reported By:                Kenneth Myers
Assigned To:                
====================================================================== 
Project:                    Asterisk
Issue ID:                   18228
Category:                   Channels/chan_sip/General
Reproducibility:            always
Severity:                   minor
Priority:                   normal
Status:                     feedback
Asterisk Version:           1.4.36 
JIRA:                        
Regression:                 No 
Reviewboard Link:            
SVN Branch (only for SVN checkouts, not tarball releases): N/A 
SVN Revision (number only!):  
Request Review:              
====================================================================== 
Date Submitted:             2010-10-28 17:02 CDT
Last Modified:              2010-11-01 14:13 CDT
====================================================================== 
Summary:                    SIP 180 Sent from provider, Asterisk sends 180 and
183 w/SDP. No ringback on calls
Description: 
When calling through a provider, the provider sends 180 to asterisk, and
asterisk sends 180 followed by 183 w/sdp to the phone. The caller hears no
ringback. See attached screenshots
====================================================================== 

---------------------------------------------------------------------- 
 (0128510) davidw (reporter) - 2010-11-01 14:13
 https://issues.asterisk.org/view.php?id=18228#c128510 
---------------------------------------------------------------------- 
This may relate to something we have observed with Cisco CUCM 8.

When connecting to the PSTN, it is sending 180 with SDP, but sending
silence (not silence suppression) in the RTP.

After signalling AST_CONTROL_RINGING, Asterisk seems to respond to the SDP
by sending AST_CONTROL_PROGRESS.  In our case, party A is already answered,
so there is no 183 to generate, but I suspecte 183 would be generated.

This is currently on our to do list to investigate, although,
provisionally I am treating it as not being an Asterisk bug.  I'm not sure
how the PSTN connectivity to the Cisco is achieved, and haven't
investigated whether there are options to change the behaviour.

RFC 3960 suggests that giving the RTP stream priority over the 180 Ringing
is reasonable behaviour, although doesn't mandate it, calling it a local
policy decision.

The pastebin version of the debugging info for this issue doesn't seem to
have any entries for the B side, so I can't tell whether this is also 180 +
SDP.

In our case, we are using a (patched) 1.6.1.0. 

Issue History 
Date Modified    Username       Field                    Change               
====================================================================== 
2010-11-01 14:13 davidw         Note Added: 0128510                          
======================================================================