[Freeswitch-users] RTP media issue

Anthony Minessale anthony.minessale at gmail.com
Fri May 25 20:06:06 MSD 2012


What version of FS are you running?
Do you have the debug logs of those calls?

you could try using the jitterbuffer.
<action application="cng_plc"/>
<action application="set" data="jitterbuffer_msec=60"/>

in the inbound DP to FS *before* you answer.

Also it looks a little odd to me in this trace if this is the same
call, it seems like you answered the call before placing the call to
the phone and that phone never answers....








On Thu, May 24, 2012 at 9:51 PM, Nathan Downes <nathandownes at hotmail.com> wrote:
> Hi,
>
> enable-soa
>
> <param name="enable-soa" value="true"/>
>
> Set the value to "false" to diable SIP SOA from sofia to tell sofia not to
> touch the exchange of SDP
>
> I don't think this is related to the exchange of an SDP message..  Can you
> elaborate more before I try it? I can't make things worse or change things I
> don't understand.
>
> ________________________________
> From: djbinter at gmail.com
> To: freeswitch-users at lists.freeswitch.org
> CC: nathan at nortec.com.au
> Subject: Re: [Freeswitch-users] RTP media issue
> Date: Fri, 25 May 2012 11:19:46 +1000
>
>
> <param name="enable-soa" value="false"/>
>
>
> Sent from my iPad
>
> On May 24, 2012, at 5:01 PM, Nathan Downes <nathandownes at hotmail.com> wrote:
>
> Hi,
>
> I had previous reported an issue with poor voice quality, appearing to stem
> from occasion wrong timestamps coming from provider, but the end user's
> experience was much worse than what I could see/hear in the trace.
>
> I have finally captured an event inbound and outbound.  The thing I don't
> understand is I thought even though FS proxied the media it didn't touch it
> or change anything, but it appears it is.
>
> The 2 traces are http://www.nortec.com.au/inbound.pcap.gz and
> http://www.nortec.com.au/outbound.pcap.gz
>
> Inbound is from my trunk provider to FS box and outbound is FS box to ATA in
> FTTH GPON.
>
> The event I am talking about, if both traces are open, is in the inbound one
> inbetween packet 8114 and 8117 the provider drops a packet or I don't
> receive it.  In the corresponding outbound trace, between packet 8144 and
> 8152,  it appears FS misses a whole heap of packets (.1 seconds) between
> 8146 and 8152 then it increases the timestamp only by 40 rather than 160 on
> packet 8152.  This seems to not affect SIP phones themselves but causes
> issues with the FTTH GPON ATA.
>
> This causes a gap in the audio for the end user, and when they miss a high
> number of packets even though it sounds good on the inbound trace the end
> users experience is horrible.   This trace is actually a good one, but the
> wrong timestamp can occur once per second, causing end user to lose 10%+ of
> incoming audio only.  The issue only affects the audio coming from provider
> to FS to end user.
>
> I am chasing it up with the voice provider to try and eliminate the
> occasional packet loss, but if I could stop/fix FS from doing its
> adjustment/gap/something the end user wouldn't even notice it.
>
>
>
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-- 
Anthony Minessale II

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