[Freeswitch-users] RTP media issue

Anthony Minessale anthony.minessale at gmail.com
Sun May 27 04:29:01 MSD 2012


Also update, you are running git head so you have to update frequently, you
rev is a month old.
On May 26, 2012 7:43 AM, "Peter Olsson" <peter.olsson at visionutveckling.se>
wrote:

> I think the first step is to check the actual signalling.
>
> Just as Anthony mentions, the call setup looks weird. Between FS and the
> ATA (if I understood the different IP's/peers correctly) it seems FS sends
> early media (183), and it takes the ATA 10 seconds to ACK this. During the
> time the call is answered on the other end, which dosn't seem to be passed
> to the ATA (probably because it's so late with the ACK - I'm not sure about
> that though). So, first check this - it seems to me that FS might still be
> in "early media" state during the entire call.
>
> About the actual RTP, I see one problem here, it's packet 8146 in
> outbound.pcap. This is a packet that is probably generated by FS (since by
> that exact time, you're missing one packet from the provider), the problem
> here is that the timestamp if way off (I'm not sure where FS gets this ts
> from). It's because of this faulty timestamp that a few packets after this
> is dropped, since FS won't send packets with a lower timestamp than before.
> This is also why packet 8152 seems to have a strange timestamp, but it's
> actually just passed from the other leg, and is the first timestamp that is
> greater then packet 8146.
>
> One possible solution would be to enable rewrite of timestamps
> (rtp-rewrite-timestamps in the sofia config), also mentioned here:
> http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio
>
> However, I think you also need to check the strangness in the SIP
> signalling.
>
> /Peter
>
> ________________________________________
> Från: freeswitch-users-bounces at lists.freeswitch.org [
> freeswitch-users-bounces at lists.freeswitch.org] f&#246;r Mr Nathan Downes [
> nathandownes at hotmail.com]
> Skickat: den 26 maj 2012 00:03
> Till: 'FreeSWITCH Users Help'
> Ämne: Re: [Freeswitch-users] RTP media issue
>
> Hi Anthony,
>
> FS version = FreeSWITCH Version 1.1.beta1 (git-f1b5044 2012-04-26 11-28-47
> -0500)
>
> I don't have a debug log, but I could probably get it with another trace of
> both sides of the call, but it would be hard to capture as there is
> constant
> calls to this, unless there is a way to do it on a per call basis? I can
> also only do testing onsite as we don't have the same fibre equipment.  I
> already have jitterbuffer set in both profiles in an attempt to try and
> stop
> it using  <param name="auto-jitterbuffer-msec" value="60"/>, is there a way
> to set the cng_plc in the profile itself rather than diaplpan as there are
> 70 or so numbers in it.  In the outbound dialplans I also added  <action
> application="set" data="sip_jitter_buffer_during_bridge=true" />  because I
> kept seeing PAUSE JITTERBUFFER in the FS logs when calls were made outbound
> so I wasn't sure it was doing something and read somewhere it pauses it
> when
> it bridges the call
>
> The inbound dialplan for all of those people consists of
>
> <extension name="02-xxxx-9365" >
>   <condition field="context" expression="public"/>
>   <condition field="destination_number" expression="^02xxxx9365;.*$">
>       <action application="sleep" data="3000"/>
>       <action application="set" data="domain=mgst.xxxxxxxx.com.au"/>
>       <action application="set" data="domain_name=${domain}"/>
>       <action application="set" data="call_direction=inbound"/>
>       <action application="transfer" data="117 XML mgst.xxxxxxxxx.com.au
> "/>
>   </condition>
> </extension>
>
> It doesn't affect SIP phones or normal ATA devices we have connected and
> only affects these FTTH GPON ATA's, but with almost 100 residents in this
> retirement village and them constantly complaining we have been given til
> Wednesday to come up with a solution or risk losing our position as the
> internet/phone provider for that retirement village.
>
> It appeared to me that what happens in that trace isn't normal behaviour, I
> did try rewriting timestamps last week, but as you suggested that appeared
> to mask the issue but not stop it from happening.  That was when I was
> losing a packet from them each second or so, which by the time it arrived
> to
> end user sounded horrible.  It has settled down a lot now and maybe 1 or 2
> packets per call, but if what is in this trace is the cause each time, that
> would explain the poor end users experience.
>
>
>
>
> -----Original Message-----
> From: freeswitch-users-bounces at lists.freeswitch.org
> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of
> Anthony
> Minessale
> Sent: Saturday, 26 May 2012 2:06 AM
> To: FreeSWITCH Users Help
> Subject: Re: [Freeswitch-users] RTP media issue
>
> 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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> >
> > 
> > 
> >
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>
>
> --
> Anthony Minessale II
>
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