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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