[SIPForum-discussion] G 723 Audio Codec Bit Rate
Andrew Yu
andrew at asiatel.com.sg
Mon Sep 3 16:15:20 UTC 2007
Hi Sreekant,
This is very interesting, if you still have the trace, please sent to me
the trace which includes both SIP/w SDP & RTP. I prefer it in .pcap
format unaltered. Thanks!
--
Cheers,
Asiatel Singapore Pte Ltd
Andrew Yu
19 Jalan Kilang Barat
#06-01, Acetech Centre
Singapore 159361
Tel: +65 6271 8233
Fax: +65 6274 4266
sreekant nair wrote:
> Hi All,
>
> Based on the inputs received from multiple people, we went ahead with
> our design. However during one of the tests we saw the problem being
> discussed.
>
> The G.723.1 codec dynamically changed its bit rate on the fly between
> two successive RTP frames. I have captured the trace file and exported
> it into a text format for easy viewing for all. Please see the portion
> highlighted in the document.
>
> Any thoughts are greatly appreciated.
>
> Thanks
>
> Sreekant Nair
>
> */Andrew Yu <andrew at asiatel.com.sg>/* wrote:
>
> it's possible to change the audio path & codec type by sending an
> reINVITE with SDP. could you paste here of the sip trace that you're
> talking about? when an RTP is in session, there is no way that you
> can
> change the codec type without an reINVITE. G.723.1 5.3kbps and 6.3
> kps
> is not inter-compatible and I believe that the SDP should have
> indicated
> this.
>
>
> sreekant nair wrote:
> > Thanks Boris,
> >
> > However, the situation is a little bit more tricky.
> >
> > It was my understanding that codec negotiation is done using the
> SDP.
> > I tried capturing the INVITE - 18X - 2XX msg using Ethereal to
> see if
> > there is anything that specifies the bit rate that will be used. In
> > all cases, the codec alone is specified. Even though G.723 supports
> > dual bit rates, I could not find anything which explicitly
> states the
> > bit rate. (At least Ethereal does not decode it so). Is there
> > something I am missing here.
> >
> > During a test scenario, both nodes negotiated and agreed on
> G.723 but
> > one node sent using 5.3kbps while the other replied with 6.3Kbps
> and
> > the audio path was established. This was found by the amount of
> data
> > bytes in the RTP packets - one way it was 20 while it was 24 in the
> > reverse direction.
> > My question was - Is it possible (both from a hardware/software
> > perspective) to change the bit rates while an RTP session is in
> > progress. So to change the bit rates, there is no need of codec
> > re-negotiation and no UPDATE / RE-INVITE would be sent. Hence my
> > confusion and requirement.
> >
> > Not sure if I made myself clear.
> >
> > Sreekant
> >
> > ----- Original Message ----
> > From: Boris vercher
> > To: sreekant nair ; discussion at sipforum.org
> > Sent: Friday, July 27, 2007 10:00:56 AM
> > Subject: RE: [SIPForum-discussion] G 723 Audio Codec Bit Rate
> >
> > No it’s not possible , if there are no codec renegotiation
> >
> >
> >
> > There are no compatibility between this two compressions
> >
> >
> >
> > Vercher Boris
> >
> >
> >
> >
> ------------------------------------------------------------------------
> >
> > *De :* discussion-bounces at sipforum.org
> > [mailto:discussion-bounces at sipforum.org] *De la part de*
> sreekant nair
> > *Envoyé :* vendredi 27 juillet 2007 15:17
> > *À :* discussion at sipforum.org
> > *Objet :* [SIPForum-discussion] G 723 Audio Codec Bit Rate
> >
> >
> >
> > Hi All,
> >
> >
> >
> > In our system being tested, one of the codecs used is G.723
> audio codec
> >
> > G.723 has two bit rates - 5.3Kbps & 6.3 Kbps.
> >
> >
> >
> > Is is possible that during a voice call, the codec can dynamically
> > change the bit rate between 5.3Kbps or 6.3 Kbps ?
> >
> >
> >
> > Any thoughts are greatly appreciated.
> >
> >
> >
> > Thanks
> >
> > Sreekant Nair
> >
> >
> >
> >
> >
> >
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> --
> Cheers,
>
> Asiatel Singapore Pte Ltd
> Andrew Yu
>
> 19 Jalan Kilang Barat
> #06-01, Acetech Centre
> Singapore 159361
>
> Tel: +65 6271 8233
> Fax: +65 6274 4266
>
>
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