-----------------------------------------------------------
This is an automatically generated e-mail. To reply, visit:
https://reviewboard.asterisk.org/r/3256/
-----------------------------------------------------------

(Updated Feb. 27, 2014, 12:41 p.m.)


Review request for Asterisk Developers, Joshua Colp, Kevin Harwell, and Matt 
Jordan.


Changes
-------

Address Josh's RTCP stun request finding


Bugs: ASTERISK-22911 and ASTERISK-23213
    https://issues.asterisk.org/jira/browse/ASTERISK-22911
    https://issues.asterisk.org/jira/browse/ASTERISK-23213


Repository: Asterisk


Description
-------

Let me start by saying this is almost certainly not the complete answer to 
solving these problems. This patch is simply an alternative to backing out the 
patch from r405234 and leaving the existing aborts in place. I've written a 
test to make sure the new patch (and likely a later patch which can resolve 
these problems with ICE more comprehensively) does not crash Asterisk and that 
can be view here: https://reviewboard.asterisk.org/r/3255/

In my reproduction of this regression, I noticed a few things. The first was 
that when starting a call with ICE from SIPML5 that Asterisk would not be able 
to full initialize the ICE session. Instead when creating the candidate 
checklist via pj_ice_sess_create_check_list, PJNATH would be unable to 
associate the srflx candidates with any host pairs when pruning the checklist. 
The SRFLX candidates would have the addresses used internally on my LAN while 
the addresses it would be matched up against would mirror those of my external 
IP (in other words, they just didn't match by address). The overall return from 
pj_ice_sess_create_check_list would be PJNATH_EICENOHOSTCAND and while the ICE 
session wouldn't get flagged as started, I still continued to receive two way 
audio on both ends in Asterisk 11.7

Asterisk 11.8 however would clear the candidate lists at this point and I would 
get one way audio instead.

The crash in 11.7 from holding was caused because upon holding the call, the 
SDP would contain the same list of ICE candidates, so Asterisk would attempt to 
start the ICE session again. This time when it entered the create check list 
function, the maximum ICE candidates in the session would be exhausted causing 
PJNATH to assert and abort (which visibly appears more or less the same as a 
crash). This work around patch checks to see if a create checklist call will 
cause that value to expand above the maximum and if it would, we abandon 
starting ICE back up and we clear the current candidate list on the ICE 
session. In my own tests this seemed to work quite well (I had two way audio, 
Asterisk didn't terminate suddenly, and I was able to hold and resume the call 
while still maintaining two way audio and music on hold for all expected ends). 
According to Vytis though, neither this patch nor the original patch to resolve 
ASTERISK-22911 by kharwell actually fixed his audio problems anyway. 
 Kharwell's patch fixed the assertions that were occurring because the 
candidate list would be cleared on any error including the 
PJNATH_EICENOHOSTCAND which most of the people in ASTERISK-23213 were most 
likely experiencing as well. I draw that conclusion because when they used this 
patch they reported that it fixed their issues without introducing any new 
crashes.

At this point, I'm not entirely sure that audio actually should be allowed to 
work when building the ICE session check list fails like it has been in this 
scenario. But we need to finalize a new Asterisk 11 release soon and we can't 
have this apparent regression in when we do so. In the future we need to find 
out more comprehensively why pj_ice_sess_create_check_list is failing and how 
to fix that.


Props to Lott Caskey for providing some improvements to the patch that I wrote.


Diffs (updated)
-----

  /branches/11/res/res_rtp_asterisk.c 408284 

Diff: https://reviewboard.asterisk.org/r/3256/diff/


Testing
-------

Tested calls and holds with a SIPML5 call to a desk phone. Confirmed two way 
audio and music on hold.
Requested that the reporter and participants of ASTERISK-23213 test the patch. 
Everyone who tested it confirmed that it fixed their problems.
Ran https://reviewboard.asterisk.org/r/3255/ and checked the log files to see 
what was happening


Thanks,

Jonathan Rose

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --

asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-dev

Reply via email to