[Freeswitch-users] RTP media issue

[Nathan Downes]

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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