[SIPForum-discussion] RTP packet loss

[Aniella Juverdeanu]

Yes, AudioCodes SBC – we do not use it – just read their white paper

thanks

From: Rajesh Kumar Talluri [mailto:rajeshsources at gmail.com]
Sent: February 19, 2015 06:39 PM
To: Aniella Juverdeanu
Cc: Raghul Prasanna; discussion at sipforum.org
Subject: Re: [SIPForum-discussion] RTP packet loss


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<mailto:Aniella.Juverdeanu at telus.com>> wrote:
>
> Hi,
>
>
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> 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
>
>
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> Thanks
>
> Aniella
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>
>
> From: discussion-bounces at sipforum.org<mailto:discussion-bounces at sipforum.org> [mailto: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<mailto:discussion at sipforum.org>
> Subject: [SIPForum-discussion] RTP packet loss
>
>
>
> HI All,
>
>
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> 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.
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> MTR traces, I am unable to rely on this as core routers generally tend to discard these ICMP packets giving us distorted results.
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> Any thoughts?
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> Thanks
>
>
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