| Summary: | ASTERISK-14365: SIP clients and "internal_timing" not working when silence suppression enabled | ||
| Reporter: | Bryan Field-Elliot (bryanfe) | Labels: | |
| Date Opened: | 2009-06-23 16:05:19 | Date Closed: | 2011-06-07 14:08:19 |
| Priority: | Minor | Regression? | No |
| Status: | Closed/Complete | Components: | Channels/chan_sip/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| Description: | Asterisk is not behaving as expected (based upon what documentation I can read) when SIP clients have "Silence Suppression" turned on and we are using the internal timer. Several apps in Asterisk, including Music on Hold, won't send any audio to the SIP client, unless the SIP client itself sends audio to Asterisk (which it won't do if Silence Suppression is enabled and the caller is quiet). Silence Suppression is important for our application because the caller will be mostly quiet (listening, and occasionally entering touchtones), and we can conserve a lot of bandwidth and support more users if we enable Silence Suppression on the ATA. My research indicates that if Asterisk is configured to use an internal timer (such as provided by dahdi_dummy), then this problem should go away, and SIP clients with Silence Suppression will function correctly. However, we think we've done everything right in terms of setup, and it still isn't working. Here are the relevant data points: - Asterisk version is 1.6.1 (Revision 202749 from SVN) - Dahdi (version 2.2.0 rc2) is installed and running. Linux module "dahdi_dummy" is loaded. No other dahdi modules are loaded. - Command-line tool "dahdi_test" returns +99% accuracy - res_timing_dahdi.so is loaded in Asterisk "modules.conf" file - "internal_timing" is set to "yes" in Asterisk "asterisk.conf" file - Asterisk CLI command "timing test" works, with reports such as: - SIP ATA's are Linksys PAP2Ts and Sipura 2102s. Attempting to test a timer with 50 ticks per second. Using the 'DAHDI' timing module for this test. It has been 1019 milliseconds, and we got 51 timer ticks Based upon my understanding of things, all of the above points to "go" with respect to proper support of SIP Silence Suppression in the client, but we're just not seeing it. Symptoms are easily reproducible. When sending audio to the client and silence suppresion is on, and the client is very quiet, then no audio is sent. If the client blows into the mic or has any other background or foreground noise, then Asterisk will continue to send it audio. | ||
| Comments: | By: Paul Belanger (pabelanger) 2010-06-02 13:37:02 Could try the latest 1.6.2 branch. There was some recent changes to timing and curiouss if this is still an issue. --- Per the Asterisk maintenance timeline page at http://www.asterisk.org/asterisk-versions maintenance (bug) support for the 1.6.0 and 1.6.1 branches has ended. For continued maintenance support please move to the 1.6.2 branch. More information on this change can be found in the release announcement: http://www.asterisk.org/node/49924 By: Paul Belanger (pabelanger) 2010-06-10 15:09:32 Suspended due to lack of activity. Please request a bug marshal in #asterisk-bugs on the IRC network irc.freenode.net to reopen the issue should you have the additional information requested. Further information can be found at http://www.asterisk.org/developers/bug-guidelines | ||