Hello, I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using: <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/> What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A. I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well. I'm missing something ? Silviu
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated. Are you rewriting timestamps or something too? On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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Hello Brian, Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps): <action application="set" data="rtp_manual_rtp_bugs=sonus_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs=send_linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=start_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=never_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs=ignore_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/> Regarding your advices: 1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ? What I understand now is that I need to : 1. <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p} Btw what means 1p ? Silviu On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
_________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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every one of those set lines resets that variable so only the last one is doing anything. you can set those with | seperator On Tue, Feb 21, 2017 at 10:09 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Brian,
Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps ):
<action application="set" data="rtp_manual_rtp_bugs=sonus_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs=send_linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=start_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=never_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs=ignore_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/>
Regarding your advices:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p
I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ?
What I understand now is that I need to :
1. <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Btw what means 1p ?
Silviu
On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
_________________________________________________________________________ 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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--
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*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Allison prompts for FreeSWITCH:
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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Hello, Thanks for feedback, I removed all of them and seems the same. So might be related to the way I set the jitter buffer.. Hope Brian can clarify if I understood correctly that I should do: 1. <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p} Silviu On Tue, Feb 21, 2017 at 5:18 PM, Michael Jerris <mike@jerris.com> wrote:
every one of those set lines resets that variable so only the last one is doing anything. you can set those with | seperator
On Tue, Feb 21, 2017 at 10:09 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Brian,
Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps):
<action application="set" data="rtp_manual_rtp_bugs= sonus_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs= send_linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=start_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=never_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs= ignore_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/>
Regarding your advices:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p
I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ?
What I understand now is that I need to :
1. <action application="set" data="rtp_jitter_buffer_ during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Btw what means 1p ?
Silviu
On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
_________________________________________________________________________ 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
--
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*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Allison prompts for FreeSWITCH:
*https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch>
Wish to schedule a meeting?
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*T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 <+1%20918-420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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_________________________________________________________________________ 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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Also you probably don't need all those RTP bug behaviors, that combo is for a severe Sonus infection. On Tue, Feb 21, 2017 at 9:31 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
Thanks for feedback, I removed all of them and seems the same. So might be related to the way I set the jitter buffer..
Hope Brian can clarify if I understood correctly that I should do:
1. <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Silviu
On Tue, Feb 21, 2017 at 5:18 PM, Michael Jerris <mike@jerris.com> wrote:
every one of those set lines resets that variable so only the last one is doing anything. you can set those with | seperator
On Tue, Feb 21, 2017 at 10:09 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Brian,
Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps):
<action application="set" data="rtp_manual_rtp_bugs=sonu s_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs=send _linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=start_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=never_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs=igno re_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=acce pt_any_packets"/>
Regarding your advices:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p
I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ?
What I understand now is that I need to :
1. <action application="set" data="rtp_jitter_buffer_during _bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Btw what means 1p ?
Silviu
On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_mse c=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
____________________________________________________________ _____________ 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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--
*Brian West* brian@freeswitch.org
*Twitter: @FreeSWITCH , @briankwest*
http://www.freeswitchbook.com http://www.freeswitchcookbook.com
Allison prompts for FreeSWITCH:
*https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch>
Wish to schedule a meeting?
http://app.timebridge.com/#/meet/freeswitch
Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch>
*T:*+19184209001 <+1%20918-420-9001> | *F:*+19184209002 <+1%20918-420-9002> | *M:*+1918424WEST (9378) *Skype:*briankwest
____________________________________________________________ _____________ 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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____________________________________________________________ _____________ 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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Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com
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I removed all that sonus stuffs, and added in the bridge string : {jitterbuffer_msec=1p,rtp_jitter_buffer_during_bridge=true,sip_secure_media=true,sdp_m_per_ptime=true,hangup_after_bridge=true} But still the same: in leg A I see a jitter of max 2 ms and avg of 1 and on leg B 26 ms max and avg of 16. Silviu On Tue, Feb 21, 2017 at 5:47 PM, Brian West <brian@freeswitch.org> wrote:
Also you probably don't need all those RTP bug behaviors, that combo is for a severe Sonus infection.
On Tue, Feb 21, 2017 at 9:31 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
Thanks for feedback, I removed all of them and seems the same. So might be related to the way I set the jitter buffer..
Hope Brian can clarify if I understood correctly that I should do:
1. <action application="set" data="rtp_jitter_buffer_during _bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Silviu
On Tue, Feb 21, 2017 at 5:18 PM, Michael Jerris <mike@jerris.com> wrote:
every one of those set lines resets that variable so only the last one is doing anything. you can set those with | seperator
On Tue, Feb 21, 2017 at 10:09 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Brian,
Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps):
<action application="set" data="rtp_manual_rtp_bugs=sonu s_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs=send _linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=star t_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=neve r_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs=igno re_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=acce pt_any_packets"/>
Regarding your advices:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p
I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ?
What I understand now is that I need to :
1. <action application="set" data="rtp_jitter_buffer_during _bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Btw what means 1p ?
Silviu
On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_mse c=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
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Seems a problem with FS I think . I tested same scenarios with FS 1.2 last stable and seems working fine. Problem seems more visible when transcoding iSAC into PCMU, and performing much better when you have ILBC to PCMU Silviu On Wed, Feb 22, 2017 at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
I removed all that sonus stuffs, and added in the bridge string : {jitterbuffer_msec=1p,rtp_jitter_buffer_during_bridge= true,sip_secure_media=true,sdp_m_per_ptime=true,hangup_after_bridge=true}
But still the same: in leg A I see a jitter of max 2 ms and avg of 1 and on leg B 26 ms max and avg of 16.
Silviu
On Tue, Feb 21, 2017 at 5:47 PM, Brian West <brian@freeswitch.org> wrote:
Also you probably don't need all those RTP bug behaviors, that combo is for a severe Sonus infection.
On Tue, Feb 21, 2017 at 9:31 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
Thanks for feedback, I removed all of them and seems the same. So might be related to the way I set the jitter buffer..
Hope Brian can clarify if I understood correctly that I should do:
1. <action application="set" data="rtp_jitter_buffer_during _bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Silviu
On Tue, Feb 21, 2017 at 5:18 PM, Michael Jerris <mike@jerris.com> wrote:
every one of those set lines resets that variable so only the last one is doing anything. you can set those with | seperator
On Tue, Feb 21, 2017 at 10:09 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Brian,
Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps):
<action application="set" data="rtp_manual_rtp_bugs=sonu s_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs=send _linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=star t_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=neve r_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs=igno re_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=acce pt_any_packets"/>
Regarding your advices:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p
I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ?
What I understand now is that I need to :
1. <action application="set" data="rtp_jitter_buffer_during _bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Btw what means 1p ?
Silviu
On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during _bridge=true"/> <action application="export" data="nolocal:jitterbuffer_mse c=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
____________________________________________________________ _____________ Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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typically you do NOT want a jitterbuffer in the bridge. How bad is the jitter coming out the b leg and how exactly are you measuring it? On Wed, Feb 22, 2017 at 10:11 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Seems a problem with FS I think . I tested same scenarios with FS 1.2 last stable and seems working fine. Problem seems more visible when transcoding iSAC into PCMU, and performing much better when you have ILBC to PCMU
Silviu
On Wed, Feb 22, 2017 at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
I removed all that sonus stuffs, and added in the bridge string : {jitterbuffer_msec=1p,rtp_jitter_buffer_during_bridge=true,sip_secure_media=true,sdp_m_per_ptime=true,hangup_after_bridge=true}
But still the same: in leg A I see a jitter of max 2 ms and avg of 1 and on leg B 26 ms max and avg of 16.
Silviu
On Tue, Feb 21, 2017 at 5:47 PM, Brian West <brian@freeswitch.org> wrote:
Also you probably don't need all those RTP bug behaviors, that combo is for a severe Sonus infection.
On Tue, Feb 21, 2017 at 9:31 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
Thanks for feedback, I removed all of them and seems the same. So might be related to the way I set the jitter buffer..
Hope Brian can clarify if I understood correctly that I should do:
1. <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Silviu
On Tue, Feb 21, 2017 at 5:18 PM, Michael Jerris <mike@jerris.com> wrote:
every one of those set lines resets that variable so only the last one is doing anything. you can set those with | seperator
On Tue, Feb 21, 2017 at 10:09 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Brian,
Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps ):
<action application="set" data="rtp_manual_rtp_bugs=sonus_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs=send_linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=start_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=never_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs=ignore_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/>
Regarding your advices:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p
I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ?
What I understand now is that I need to :
1. <action application="set" data="rtp_jitter_buffer_during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Btw what means 1p ?
Silviu
On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
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Hello Michael, I want a jitterbuffer in the bridge because some PSTN gateways sucks in case they receive the calls with jitter. When you have a server to server connection it's easy to control jitter by putting some dedicated links but you can't control the jitter sent by clients around the globe. I'm looking with wireshark with RTP stream analyse to measure it. And regarding how bad is on leg B, for example with iSAC on leg a I see 2 ms max and 1 ms average and on leg B for the same call 18 ms avg and 26 max. Silviu On Wed, Feb 22, 2017 at 5:52 PM, Michael Jerris <mike@jerris.com> wrote:
typically you do NOT want a jitterbuffer in the bridge. How bad is the jitter coming out the b leg and how exactly are you measuring it?
On Wed, Feb 22, 2017 at 10:11 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Seems a problem with FS I think . I tested same scenarios with FS 1.2 last stable and seems working fine. Problem seems more visible when transcoding iSAC into PCMU, and performing much better when you have ILBC to PCMU
Silviu
On Wed, Feb 22, 2017 at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
I removed all that sonus stuffs, and added in the bridge string : {jitterbuffer_msec=1p,rtp_jitter_buffer_during_bridge= true,sip_secure_media=true,sdp_m_per_ptime=true,hangup_after_bridge=true}
But still the same: in leg A I see a jitter of max 2 ms and avg of 1 and on leg B 26 ms max and avg of 16.
Silviu
On Tue, Feb 21, 2017 at 5:47 PM, Brian West <brian@freeswitch.org> wrote:
Also you probably don't need all those RTP bug behaviors, that combo is for a severe Sonus infection.
On Tue, Feb 21, 2017 at 9:31 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
Thanks for feedback, I removed all of them and seems the same. So might be related to the way I set the jitter buffer..
Hope Brian can clarify if I understood correctly that I should do:
1. <action application="set" data="rtp_jitter_buffer_ during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Silviu
On Tue, Feb 21, 2017 at 5:18 PM, Michael Jerris <mike@jerris.com> wrote:
every one of those set lines resets that variable so only the last one is doing anything. you can set those with | seperator
On Tue, Feb 21, 2017 at 10:09 AM Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Brian,
Thanks for your feedback. I don't think I have something to rewrite timestamps. I also use the following (hope none of is rewriting timestamps):
<action application="set" data="rtp_manual_rtp_bugs= sonus_send_invalid_timestamp_2833"/> <action application="set" data="rtp_manual_rtp_bugs= send_linear_timestamps"/> <action application="set" data="rtp_manual_rtp_bugs=start_seq_at_zero"/> <action application="set" data="rtp_manual_rtp_bugs=never_send_marker"/> <action application="set" data="rtp_manual_rtp_bugs= ignore_dtmf_duration"/> <action application="set" data="rtp_manual_rtp_bugs=accept_any_packets"/>
Regarding your advices:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p
I'm not sure is clear enough for me. Can you please past here how looks this settings in the bridge line ? or where I can find an example in documentation ?
What I understand now is that I need to :
1. <action application="set" data="rtp_jitter_buffer_ during_bridge=true"/> 2. And in bridge string add : {jitterbuffer_msec=1p}
Btw what means 1p ?
Silviu
On Tue, Feb 21, 2017 at 4:43 PM, Brian West <brian@freeswitch.org> wrote:
1. Don't use the nolocal export method, set those variables in side the {} on the bridge line. 2. set jitterbuffer_msec=1p 3. The jitter buffer was ripped and rebuilt so some of the documentation hasn't been updated.
Are you rewriting timestamps or something too?
On Tue, Feb 21, 2017 at 8:26 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
I'm using last FS 1.6.15 to bridge a call and to transcode it as well from ILBC/ISAC to PCMU. To enable jitter buffer during bridge I'm using:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/> <action application="export" data="nolocal:jitterbuffer_msec=60:200:20"/>
What I observed is that looking with wireshark still shows jitter on leg B which is sent to PSTN even if there is no jitter on leg A, or shows worst jitter on leg B in case I have jitter on call A.
I was expected that the jitter buffer will fix this and even if I have a slightly low jitter on leg A will not be propagated on le B as well.
I'm missing something ?
Silviu
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Hello Russel, Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter. Silviu On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven <rtreleaven@bunnykick.ca> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in
case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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I could be way out to lunch here but shouldn't the jitter buffer be on the a-leg? On Wed, Feb 22, 2017 at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in
case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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-- Sincerely, Russell Treleaven sip:rtreleaven@sip.bunnykick.ca
what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca>> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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Hello michael, http://www.digitalk.com/ is the one generating problems where I can replicate problem all the time. Silviu On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com> wrote:
what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in
case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello michael,
http://www.digitalk.com/ <http://www.digitalk.com/> is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com <mailto:silviu.cpp@gmail.com>> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca>> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter... Silviu On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com> wrote:
any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello michael,
http://www.digitalk.com/ is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com> wrote:
what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in
case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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based on the numbers you are seeing, it does not sound like jitter is actually your issue. I’d look closer at the numbers, my guess is you see one weird timestamp in there and it has the jitter calculation messed up.
On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter...
Silviu
On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com <mailto:silviu.cpp@gmail.com>> wrote:
Hello michael,
http://www.digitalk.com/ <http://www.digitalk.com/> is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com <mailto:silviu.cpp@gmail.com>> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca>> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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But do you have any idea what can breaks the timestamps ? Because this takes place with both ISAC and ILBC Any idea in what direction I can debug ? Silviu On Wed, Feb 22, 2017 at 11:22 PM, Michael Jerris <mike@jerris.com> wrote:
based on the numbers you are seeing, it does not sound like jitter is actually your issue. I’d look closer at the numbers, my guess is you see one weird timestamp in there and it has the jitter calculation messed up.
On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter...
Silviu
On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com> wrote:
any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello michael,
http://www.digitalk.com/ is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com> wrote:
what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in
case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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Without a debug log with sip trace, no… could be different ptimes on each leg?
On Feb 22, 2017, at 4:36 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
But do you have any idea what can breaks the timestamps ? Because this takes place with both ISAC and ILBC Any idea in what direction I can debug ?
Silviu
On Wed, Feb 22, 2017 at 11:22 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: based on the numbers you are seeing, it does not sound like jitter is actually your issue. I’d look closer at the numbers, my guess is you see one weird timestamp in there and it has the jitter calculation messed up.
On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp@gmail.com <mailto:silviu.cpp@gmail.com>> wrote:
No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter...
Silviu
On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com <mailto:silviu.cpp@gmail.com>> wrote:
Hello michael,
http://www.digitalk.com/ <http://www.digitalk.com/> is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com <mailto:mike@jerris.com>> wrote: what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com <mailto:silviu.cpp@gmail.com>> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven <rtreleaven@bunnykick.ca <mailto:rtreleaven@bunnykick.ca>> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks in case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
Hi Michael, It's enough the logs FS is producing with : sofia loglevel all 9 sofia global siptrace on console loglevel 7 ?? On Thu, Feb 23, 2017 at 12:06 AM, Michael Jerris <mike@jerris.com> wrote:
Without a debug log with sip trace, no… could be different ptimes on each leg?
On Feb 22, 2017, at 4:36 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
But do you have any idea what can breaks the timestamps ? Because this takes place with both ISAC and ILBC Any idea in what direction I can debug ?
Silviu
On Wed, Feb 22, 2017 at 11:22 PM, Michael Jerris <mike@jerris.com> wrote:
based on the numbers you are seeing, it does not sound like jitter is actually your issue. I’d look closer at the numbers, my guess is you see one weird timestamp in there and it has the jitter calculation messed up.
On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter...
Silviu
On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com> wrote:
any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello michael,
http://www.digitalk.com/ is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com> wrote:
what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks
in case they receive the calls with jitter.
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
Professional FreeSWITCH Consulting Services: consulting@freeswitch.org http://www.freeswitchsolutions.com
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FYI sip_secure_media became rtp_secure_media a while back. On Thu, Feb 23, 2017 at 4:14 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hi Michael,
It's enough the logs FS is producing with :
sofia loglevel all 9 sofia global siptrace on console loglevel 7
??
On Thu, Feb 23, 2017 at 12:06 AM, Michael Jerris <mike@jerris.com> wrote:
Without a debug log with sip trace, no… could be different ptimes on each leg?
On Feb 22, 2017, at 4:36 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
But do you have any idea what can breaks the timestamps ? Because this takes place with both ISAC and ILBC Any idea in what direction I can debug ?
Silviu
On Wed, Feb 22, 2017 at 11:22 PM, Michael Jerris <mike@jerris.com> wrote:
based on the numbers you are seeing, it does not sound like jitter is actually your issue. I’d look closer at the numbers, my guess is you see one weird timestamp in there and it has the jitter calculation messed up.
On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter...
Silviu
On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com> wrote:
any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello michael,
http://www.digitalk.com/ is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com> wrote:
what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
I want a jitterbuffer in the bridge because some PSTN gateways sucks > in case they receive the calls with jitter. >
How and why do they suck when they receive calls with jitter? Are we talking about all PSTN gateways or just some of them?
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-- *Brian West* brian@freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Wish to schedule a meeting? http://app.timebridge.com/#/meet/freeswitch Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest
Hello, @Brian I fixed that problem. Anyway issue still persist. I added a bug https://freeswitch.org/jira/browse/FS-10072 and also attached logs as Mike suggested Silviu On Thu, Feb 23, 2017 at 9:41 PM, Brian West <brian@freeswitch.org> wrote:
FYI sip_secure_media became rtp_secure_media a while back.
On Thu, Feb 23, 2017 at 4:14 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hi Michael,
It's enough the logs FS is producing with :
sofia loglevel all 9 sofia global siptrace on console loglevel 7
??
On Thu, Feb 23, 2017 at 12:06 AM, Michael Jerris <mike@jerris.com> wrote:
Without a debug log with sip trace, no… could be different ptimes on each leg?
On Feb 22, 2017, at 4:36 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
But do you have any idea what can breaks the timestamps ? Because this takes place with both ISAC and ILBC Any idea in what direction I can debug ?
Silviu
On Wed, Feb 22, 2017 at 11:22 PM, Michael Jerris <mike@jerris.com> wrote:
based on the numbers you are seeing, it does not sound like jitter is actually your issue. I’d look closer at the numbers, my guess is you see one weird timestamp in there and it has the jitter calculation messed up.
On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter...
Silviu
On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com> wrote:
any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello michael,
http://www.digitalk.com/ is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com> wrote:
what type of switch are you talking to when you have problems?
On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello Russel,
Not all, depends on what technologies they have behind. For example I have some calls sent via two different gateways with almost same jitter and using route A I have breaks in voice using the second one it's fine. If I send via route A without jitter quality is clear. Once you put some jitter sucks. We started to have this problem once we upgraded from FS 1.2 to 1.6 and only noticed difference is this jitter.
Silviu
On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < rtreleaven@bunnykick.ca> wrote:
> > > I want a jitterbuffer in the bridge because some PSTN gateways sucks >> in case they receive the calls with jitter. >> > > How and why do they suck when they receive calls with jitter? > Are we talking about all PSTN gateways or just some of them? > > >
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I tested now with 20bcb2edef02e95118df76f0ecdd8a64471d1e0a (last master) and works nice . There is no jitter as output Is master stable for production ? Because 1.6.x seems that has this bug, also in 1.2.x which is dead for long time it's fine Silviu On Tue, Feb 28, 2017 at 4:45 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello,
@Brian I fixed that problem.
Anyway issue still persist. I added a bug https://freeswitch.org/jira/ browse/FS-10072 and also attached logs as Mike suggested
Silviu
On Thu, Feb 23, 2017 at 9:41 PM, Brian West <brian@freeswitch.org> wrote:
FYI sip_secure_media became rtp_secure_media a while back.
On Thu, Feb 23, 2017 at 4:14 AM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hi Michael,
It's enough the logs FS is producing with :
sofia loglevel all 9 sofia global siptrace on console loglevel 7
??
On Thu, Feb 23, 2017 at 12:06 AM, Michael Jerris <mike@jerris.com> wrote:
Without a debug log with sip trace, no… could be different ptimes on each leg?
On Feb 22, 2017, at 4:36 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
But do you have any idea what can breaks the timestamps ? Because this takes place with both ISAC and ILBC Any idea in what direction I can debug ?
Silviu
On Wed, Feb 22, 2017 at 11:22 PM, Michael Jerris <mike@jerris.com> wrote:
based on the numbers you are seeing, it does not sound like jitter is actually your issue. I’d look closer at the numbers, my guess is you see one weird timestamp in there and it has the jitter calculation messed up.
On Feb 22, 2017, at 3:20 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
No, I think they have something proprietary. But anyway this problem can be with any solution that's not able to handle jitter properly. And what's killing me is why in FS I have this behaviour.. I mean after transcoding the stream has jitter... I mean I send a stream with almost no jitter and after transcoding results a stream with jitter...
Silviu
On Wed, Feb 22, 2017 at 10:12 PM, Michael Jerris <mike@jerris.com> wrote:
any info in their sdp or sip packets to say what kind of equipment they are using?
On Feb 22, 2017, at 3:00 PM, Caragea Silviu <silviu.cpp@gmail.com> wrote:
Hello michael,
http://www.digitalk.com/ is the one generating problems where I can replicate problem all the time.
Silviu
On Wed, Feb 22, 2017 at 9:34 PM, Michael Jerris <mike@jerris.com> wrote:
> what type of switch are you talking to when you have problems? > > On Feb 22, 2017, at 2:16 PM, Caragea Silviu <silviu.cpp@gmail.com> > wrote: > > Hello Russel, > > Not all, depends on what technologies they have behind. For example > I have some calls sent via two different gateways with almost same jitter > and using route A I have breaks in voice using the second one it's fine. If > I send via route A without jitter quality is clear. Once you put some > jitter sucks. We started to have this problem once we upgraded from FS 1.2 > to 1.6 and only noticed difference is this jitter. > > Silviu > > On Wed, Feb 22, 2017 at 8:48 PM, Russell Treleaven < > rtreleaven@bunnykick.ca> wrote: > >> >> >> I want a jitterbuffer in the bridge because some PSTN gateways >>> sucks in case they receive the calls with jitter. >>> >> >> How and why do they suck when they receive calls with jitter? >> Are we talking about all PSTN gateways or just some of them? >> >> >> >
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participants (4)
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Brian West
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Caragea Silviu
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Michael Jerris
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Russell Treleaven