Hello, I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102. My connections is as follows: ATA_A---->FS_A---->vpn---->FS_B---->ATA_B When user A is calling to B where should i specify jitter buffer amongst the following: 1. The bridge between Freeswitch instance FS_A and FS_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/> 2. The bridge between Freeswitch instance FS_B and ATA_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/> Thanks, Rajil
The variable must be set on the leg before the channel engages media. you can test if its on with uuid_jitterbuffer <uuid> debug:10 if you want it on both legs set the following global vars in vars.xml rtp_jitter_buffer_during_bridge=true jitterbuffer_msec=60 On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Hello,
I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102.
My connections is as follows:
ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
When user A is calling to B where should i specify jitter buffer amongst the following:
1. The bridge between Freeswitch instance FS_A and FS_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>
2. The bridge between Freeswitch instance FS_B and ATA_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>
Thanks, Rajil
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says, If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality. So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says? On 15 February 2016 at 17:10, Anthony Minessale <anthony.minessale@gmail.com
wrote:
The variable must be set on the leg before the channel engages media. you can test if its on with
uuid_jitterbuffer <uuid> debug:10
if you want it on both legs set the following global vars in vars.xml
rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60
On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Hello,
I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102.
My connections is as follows:
ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
When user A is calling to B where should i specify jitter buffer amongst the following:
1. The bridge between Freeswitch instance FS_A and FS_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>
2. The bridge between Freeswitch instance FS_B and ATA_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>
Thanks, Rajil
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end. That is why its not defaulted to being possible but you asked how to do it. On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality.
So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?
On 15 February 2016 at 17:10, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The variable must be set on the leg before the channel engages media. you can test if its on with
uuid_jitterbuffer <uuid> debug:10
if you want it on both legs set the following global vars in vars.xml
rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60
On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Hello,
I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102.
My connections is as follows:
ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
When user A is calling to B where should i specify jitter buffer amongst the following:
1. The bridge between Freeswitch instance FS_A and FS_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>
2. The bridge between Freeswitch instance FS_B and ATA_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>
Thanks, Rajil
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
Ok, what about my specific use case where both endpoints ATA (Linksys spa3102) do have internal jitterbuffer. Should i still set the global variables as you suggested to improve the call quality over vpn? On 15 February 2016 at 18:23, Anthony Minessale <anthony.minessale@gmail.com
wrote:
If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end. That is why its not defaulted to being possible but you asked how to do it.
On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality.
So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?
On 15 February 2016 at 17:10, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The variable must be set on the leg before the channel engages media. you can test if its on with
uuid_jitterbuffer <uuid> debug:10
if you want it on both legs set the following global vars in vars.xml
rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60
On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Hello,
I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102.
My connections is as follows:
ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
When user A is calling to B where should i specify jitter buffer amongst the following:
1. The bridge between Freeswitch instance FS_A and FS_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>
2. The bridge between Freeswitch instance FS_B and ATA_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>
Thanks, Rajil
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
When you keep the codec the same across the whole call path then the jitterbuffer is not as important because FS will preserve the timestamps. If you are transcoding or recording the call, then the JB can help because you are creating new timestamps so the far end can no longer perceive the jitter. Ideally you should just try it. Also you should make sure your vpn is using udp or gprs not tcp You should probably try it with everything PCMU or PCMA as well with and without the JB. Trial and error is your friend. But without the vars I mentioned it will never use the JB during a bridge. On Mon, Feb 15, 2016 at 6:33 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Ok, what about my specific use case where both endpoints ATA (Linksys spa3102) do have internal jitterbuffer. Should i still set the global variables as you suggested to improve the call quality over vpn?
On 15 February 2016 at 18:23, Anthony Minessale < anthony.minessale@gmail.com> wrote:
If one or the other end of the call does not have its own JB then putting it on both sides can fix the stream for the other end. That is why its not defaulted to being possible but you asked how to do it.
On Mon, Feb 15, 2016 at 5:54 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
The wiki page (https://wiki.freeswitch.org/wiki/Jitterbuffer) says,
If both sides of a bridge are RTP and both sides have a jb, its fairly useless. In fact if anything, it can worsen call quality.
So if I set jitterbuffer on both of my FS instances as global variables, wont that be against what the wiki says?
On 15 February 2016 at 17:10, Anthony Minessale < anthony.minessale@gmail.com> wrote:
The variable must be set on the leg before the channel engages media. you can test if its on with
uuid_jitterbuffer <uuid> debug:10
if you want it on both legs set the following global vars in vars.xml
rtp_jitter_buffer_during_bridge=true
jitterbuffer_msec=60
On Mon, Feb 15, 2016 at 4:44 PM, Rajil Saraswat <rajil.s@gmail.com> wrote:
Hello,
I have two FS instances connected to each other over VPN. The VPN sometimes has poor connectivity and I was wondering whether jitterbuffer can help to improve the call quality. I am using the OPUS codec between the FS instances since it is supposed to behave better in lower bandwidth. The ATA on both ends are Linksys SPA3102.
My connections is as follows:
ATA_A---->FS_A---->vpn---->FS_B---->ATA_B
When user A is calling to B where should i specify jitter buffer amongst the following:
1. The bridge between Freeswitch instance FS_A and FS_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@FS_B_IP"/>
2. The bridge between Freeswitch instance FS_B and ATA_B <action application="bridge" data="{jitterbuffer_msec=60}sofia/internal/$1@ATA_B_IP"/>
Thanks, Rajil
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬
☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>*
ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900
https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
FreeSWITCH-users mailing list FreeSWITCH-users@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org
-- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA
participants (2)
-
Anthony Minessale
-
Rajil Saraswat