[SIPForum-discussion] RTP packet loss

[Rajesh Kumar Talluri]

Hi Aniella,

You mean to say that Audiocodes box will check the QoS parameters on the
service provider network and provide 1-2 sec. Are you talking about
Audiocodes SBC here? Or Is it a different?

Thanks,

Raj

On Feb 19, 2015 12:56 AM, "Aniella Juverdeanu" <Aniella.Juverdeanu at telus.com>
wrote:
>
> Hi,
>
>
>
> You might need to analyze the Jitter buffer on the segment that is the
access into your network. If yours I setup too low, then RTP packets are
discarded. Usually this happens if the access is from IP networks where
there is no QOS (i.e. ISP providers). AudioCodes has a good approach to
this adaptive jitter buffer – can provide up to 1-2 sec JB
>
>
>
> Thanks
>
> Aniella
>
>
>
> From: discussion-bounces at sipforum.org [mailto:
discussion-bounces at sipforum.org] On Behalf Of Raghul Prasanna
> Sent: February 13, 2015 04:07 AM
> To: discussion at sipforum.org
> Subject: [SIPForum-discussion] RTP packet loss
>
>
>
> HI All,
>
>
>
> If there is rtp packet loss resulting in bad audio and the loss is NOT
happening on the network we control, what are the options to identify where
it's happening and possibly correct it.
>
>
>
> MTR traces, I am unable to rely on this as core routers generally tend to
discard these ICMP packets giving us distorted results.
>
>
>
> Any thoughts?
>
>
>
> Thanks
>
>
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