Hi I have updated this page with info about the new mod_opus options, including use-jb-lookahead : https://freeswitch.org/confluence/display/FREESWITCH/mod_opus Bote, the new JB in FS master applies for all audio codecs, only that FEC works only for codecs that support FEC. I know only about Opus, SILK and AMR ,but there might be others that have FEC. I have only tested Opus successfully with the new JB, the other audio codec modules might need some code changes, but I am not sure. As Anthony said, "p" comes from "number of packets". That's because the call might be established at a different ptime than 20 ms , and the JB size should be adjusted. Regards, Dragos On 11/15/2015 12:19 AM, Anthony Minessale wrote:
it can be a number of ms min and max or a number of packets min and max, max is optional.
1p:25p 20:150 1p 20
At any time after its runnint it can be debug:<level> to change debugging
On Sat, Nov 14, 2015 at 1:55 PM, Bote Man <bote_radio@botecomm.com <mailto:bote_radio@botecomm.com>> wrote:
OK, I just watched the Wednesday conference video during which Brian demonstrated the jitter buffer. I caught it live, but got in late.
Does the new jb only apply to Opus with FEC turned on? Or all codecs that support FEC? Or all codecs, period?
It would help me immensely if I had a list of current commands that are applicable to the jitter buffer, as Brian says that there are incorrect ones that need to be cleaned out. I’m only fair to middling at reading source code and figuring out what’s happening.
I saw on the video he typed “1p” at the end of a jb command and a guy on the mailing list just offered a similar construct in a thread discussing the jb, which puzzled both of us.
Thanks.
Bote
*From:* freeswitch-docs-bounces@lists.freeswitch.org <mailto:freeswitch-docs-bounces@lists.freeswitch.org> [mailto:freeswitch-docs-bounces@lists.freeswitch.org <mailto:freeswitch-docs-bounces@lists.freeswitch.org>] *On Behalf Of *Anthony Minessale *Sent:* Saturday, 14 November, 2015 10:57 *To:* FreeSWITCH Docs Team
*Subject:* Re: [Freeswitch-docs] [Freeswitch-users] Jitterbuffer
Its there under the cluecon weekly from a few weeks ago.
On Friday, November 13, 2015, Bote Man <bote_radio@botecomm.com <mailto:bote_radio@botecomm.com>> wrote:
This is a good idea, except which docs are invalid now?
I added the use-jb-lookahead under a new section about Opus and the note at the bottom of the Confluence page from ”Bipin”, but don’t know what to delete, if anything.
I looked for on YouTube for your excellent demo of the new jitter buffer performance but it’s not up there yet; I was hoping it had some of the new stuff outlined on a screen capture.
Thanks.
Bote
*From:* freeswitch-docs-bounces@lists.freeswitch.org [mailto:freeswitch-docs-bounces@lists.freeswitch.org] *On Behalf Of *Brian West *Sent:* Friday, 13 November, 2015 12:45 *To:* FreeSWITCH Users Help; FreeSWITCH Docs Team *Subject:* Re: [Freeswitch-docs] [Freeswitch-users] Jitterbuffer
We probably need to wipe confluence clean of the incorrect jitter buffer docs now that its been completely rewritten in the past few months.
On Thu, Nov 12, 2015 at 4:06 AM, Ola Bäckström <Ola.Backstrom@loxysoft.se> wrote:
Dragos,
Your example to enable jitter buffering uses the letter ”p”. Is that a typo or does it indeed mean something?
> You can enable the Jitter Buffer in the dial plan: > <action application="set" data="jitterbuffer_msec=2p:25p:"/>
I see no such usage in the documentation at https://freeswitch.org/confluence/display/FREESWITCH/JitterBuffer , but just “60:200:20”.
Best regards
/Ola
*From:* freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of *Dragos Oancea *Sent:* den 12 november 2015 10:14 *To:* freeswitch-users@lists.freeswitch.org *Subject:* Re: [Freeswitch-users] Jitterbuffer
Hi Josh,
Use FS master, it has a new audio Jitter Buffer.
Then you can use Opus with this setting in the codec config: <param name="use-jb-lookahead" value="1"/> . The other side must be sending FEC so you could profit from it.
You can enable the Jitter Buffer in the dial plan: <action application="set" data="jitterbuffer_msec=2p:25p:"/> And then enable PLC : <action application="set" data="rtp_jitter_buffer_plc=true"/>
Regards, Dragos
On 11/12/2015 07:51 AM, Josh Richesin wrote:
Hello,
I am running FreeSwitch for our small Wireless ISP. Obviously, as a wireless ISP and ITSP, we have a lot of jitter and some packet loss. Have a few questions for this setup.
1) What is the best codec to use when there is going to be jitter and some packetloss?
2) The ATA's that we are using have adaptive jitter buffers, should I enable jitterbuffer on Freeswitch as well? If so, should it be via sofia profiles?
The problem that I am trying to solve is this...
Our voice customers on our freeswitch have a lot of issues with "calls breaking up" When they use another voice provider, like say vonage or something running off of a metaswitch, they do not have the same issues.
Any help with this would be appreciated! Thanks in advance.
Josh
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