Dear All, We have developed our own sip application that works with Freeswitch. So, on incoming rtp stream we can ajust jitter buffer on client side, but on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a way to switch Jitter Buffer only for outgoing stream in Freeswitch Thanks -- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin
Jitter buffers only work on inbound media streams. Perhaps you need to rephrase? Do you mean to run a jitter buffer on the outbound leg of a call but presumably on the inbound media stream of that leg. If so, you just set the variable jitterbuffer_msec to the desired starting size some time before the channel is answered. On Wed, Jun 3, 2015 at 6:09 AM, Kamil Nigmatullin < kamil.nigmatullin@gmail.com> wrote:
Dear All,
We have developed our own sip application that works with Freeswitch. So, on incoming rtp stream we can ajust jitter buffer on client side, but on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a way to switch Jitter Buffer only for outgoing stream in Freeswitch
Thanks
-- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin
_________________________________________________________________________ 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
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-- 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
you don't buffer outgoing rtp. jitter is introduced on the wire after it leaves the box so buffering outgoing streams would only delay audio and not add any value. On Wednesday, June 3, 2015, Kamil Nigmatullin <kamil.nigmatullin@gmail.com> wrote:
Dear All,
We have developed our own sip application that works with Freeswitch. So, on incoming rtp stream we can ajust jitter buffer on client side, but on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a way to switch Jitter Buffer only for outgoing stream in Freeswitch
Thanks
-- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin
What I mean, I have this scheme: [Android Phone on 3G] <---> [Freeswitch] <---->[DID Provider]. When Android user calls DID Provider, I want to apply Jitter Buffer on Freeswitch so that voice (stream) coming from Android become smoother. On the same time, I want that the stream from DID provider of the same call was not affected by Jitter Buffer and will be delivered to Android Phone unchanged. Thanks 2015-06-03 21:09 GMT+06:00 Michael Jerris <mike@jerris.com>:
you don't buffer outgoing rtp. jitter is introduced on the wire after it leaves the box so buffering outgoing streams would only delay audio and not add any value.
On Wednesday, June 3, 2015, Kamil Nigmatullin <kamil.nigmatullin@gmail.com> wrote:
Dear All,
We have developed our own sip application that works with Freeswitch. So, on incoming rtp stream we can ajust jitter buffer on client side, but on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a way to switch Jitter Buffer only for outgoing stream in Freeswitch
Thanks
-- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin
_________________________________________________________________________ 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
-- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin
Enable jitter buffer at 20 ms on both legs and call it a day, its adaptive. On Wednesday, June 3, 2015, Kamil Nigmatullin <kamil.nigmatullin@gmail.com> wrote:
What I mean, I have this scheme: [Android Phone on 3G] <---> [Freeswitch] <---->[DID Provider]. When Android user calls DID Provider, I want to apply Jitter Buffer on Freeswitch so that voice (stream) coming from Android become smoother. On the same time, I want that the stream from DID provider of the same call was not affected by Jitter Buffer and will be delivered to Android Phone unchanged.
Thanks
2015-06-03 21:09 GMT+06:00 Michael Jerris <mike@jerris.com <javascript:_e(%7B%7D,'cvml','mike@jerris.com');>>:
you don't buffer outgoing rtp. jitter is introduced on the wire after it leaves the box so buffering outgoing streams would only delay audio and not add any value.
On Wednesday, June 3, 2015, Kamil Nigmatullin < kamil.nigmatullin@gmail.com <javascript:_e(%7B%7D,'cvml','kamil.nigmatullin@gmail.com');>> wrote:
Dear All,
We have developed our own sip application that works with Freeswitch. So, on incoming rtp stream we can ajust jitter buffer on client side, but on outgoing stream I want freeswitch to manage Jitter Buffer. Is there a way to switch Jitter Buffer only for outgoing stream in Freeswitch
Thanks
-- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org <javascript:_e(%7B%7D,'cvml','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 <javascript:_e(%7B%7D,'cvml','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
-- Kamil Nigmatullin Tel: 77272323748 mob: 7 (707) 2517003 Skype: kamil.nigmatullin
-- 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 (3)
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Anthony Minessale
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Kamil Nigmatullin
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Michael Jerris