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Summary:ASTERISK-06493: hide/show silenceSupp=off on SDP header
Reporter:Eliel Sardanons (eliel)Labels:
Date Opened:2006-03-07 10:16:39.000-0600Date Closed:2006-03-08 10:05:06.000-0600
Priority:MajorRegression?No
Status:Closed/CompleteComponents:Channels/chan_sip/Interoperability
Versions:Frequency of
Occurrence
Related
Issues:
Environment:Attachments:( 0) sip_silencesupp_cli_man.patch
( 1) sip_silencesupp.patch
Description:While trying to use a huawei softswitch, we notice that they have problems with our SDP package header. What we have done was strip from the header the line "a=SilenceSupp:off" and that way things started working right. To accomplish this we commented this line from channels/chan_sip.c:
/* ast_build_string(&a_audio_next, &a_audio_left, "a=silenceSupp:off - - - - -\r\n");  */

This is a patch to solve this "issue", now you can setup on sip.conf on each peer "silencesupp=hide/show" default is 'show'.
I don't know if this is the right way to solve this problem. Please check it out...

Thanks

****** ADDITIONAL INFORMATION ******

Here is the patch....
Comments:By: Olle Johansson (oej) 2006-03-07 10:23:12.000-0600

You should propably use a SIP_PAGE2 flag instead. Also make sure it's shown in "sip show peers" etc.

By: Russell Bryant (russell) 2006-03-07 11:32:47.000-0600

I really don't think we should have to introduce extra options because of other broken implementations.  It seems like this bug report would be more appropriately filed with Huawei.

By: Olle Johansson (oej) 2006-03-07 11:54:32.000-0600

According to e-mails in asterisk-dev, there are more broken implementations out there... I am very hesitant about this too.

By: Eliel Sardanons (eliel) 2006-03-07 12:23:12.000-0600

New patch using SIP_PAGE2 and showing on sip show peers and the manager the status.

By: Eliel Sardanons (eliel) 2006-03-07 12:32:41.000-0600

I don't know if it's important to show the status on "sip show peers", or on the manager... This is a trivial config parameter and doesn't affect other connections than the bogus ones... (Just to continue with a clean CLI output)

By: Olle Johansson (oej) 2006-03-07 14:31:35.000-0600

With the patch in ASTERISK-5230 this won't be a problem anymore, so I prefer to go that route. It does require a zaptel timer, but actually does the proper thing. Just removing this line without being able to handle incoming silence suppression is asking for other problems.

Will close this bug report, but leave it in archive for people that needs it to communicate with broken devices.

By: Eliel Sardanons (eliel) 2006-03-08 06:24:02.000-0600

Ok!, so, you will apply part/all the patch from ASTERISK-5230? or you will leave every thing like now.

By: Olle Johansson (oej) 2006-03-08 06:46:11.000-0600

We will have to test 5374 a bit more, so at this point nothing goes into svn. I think this patch solves your problem for now, even though it won't be part of Asterisk.