Hi,

Isn't this a problem for every audio stream. The RTP player has a 
configurable jitter buffer, but the actual endpoint usually has a 
dynamic one. Also packet loss concealment can improve the perception at 
the endpoint considerably.
So the RTP player is nice, but isn't really relevant for the end user 
perception.

Thanx,
Jaap

Luis EG Ontanon wrote:
> Just note:
> Most AMR codec/decoders take all three streams and choose the best one
> they can use but what you would hear as a middleman might not be what
> someone in the other side of a radio link is hearing. That is if we
> feed a codec in the middle we probably feed them good frames and it
> will use the higher quality stream codec (usu 14.4kbps) but the actual
> receiver might be using another lower quality stream (hearing flutes
> when listening to violins) because the radio link does not provide
> enough bandwidth for the high quality one.
> 
> That means listening to the AMR that passes through a Iu/Nb link would
> be useful for snooping a conversation but non useful for diagnostic
> purposes as you might not hear the same audio the other folk gets.
> 
> On Nov 30, 2007 4:43 PM, Anders Broman <[EMAIL PROTECTED]> wrote:
>>
>> Hi,
>> AMR is a licensed codec so it can't be provided with Wireshark but a plugin
>> interface exists to add your own codecs you'll have to do it yourself
>> though.
>> See also some other recent posts with AMR in the subject line.
>> Regards
>> Anders
>>
>>  ________________________________
>>
>> From: [EMAIL PROTECTED]
>> [mailto:[EMAIL PROTECTED] On Behalf Of Sanghee Lee
>> Sent: den 30 november 2007 01:47
>>
>> To: wireshark-dev@wireshark.org
>> Subject: [Wireshark-dev] How to playback the AMR in RTP packet
>>
>>
>> Hi,
>>
>> I know "VOIP calls" in Wireshark for playing the RTP packet.
>>
>> But, the VOIP calls only can play the G.711. I'd like to play the AMR
>> payload.
>>
>>
>>
>> Thanks in advance,
>>
>> Leo

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