[cisco-voip] When is Transcoding Required?

Fri Dec 11 01:22:58 EST 2009

Very good explanation.

Thanks!

Aaron.

From: admin at danofive.id.au [mailto:admin at danofive.id.au] On Behalf Of
Daniel
Sent: Friday, 11 December 2009 12:30 PM
To: Aaron Riemer
Cc: cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] When is Transcoding Required?

Yes you can selectively configure what codec is used between phones,
gateways and sites etc.... You can do this by using regions, also on
H323 dial peers you can set the codec.

Regions

Codecs can be configured between regions so that branch talking to core
over the WAN is a a G729 call between regions, you would set the service
parameter so that G729 is default between regions (inter region) and
then default that its G711 within a region (Intra region). You can then
manually configure specific regions, for example we use G711 to a Music
on Hold region so that we can do G711 MoH from a router flash and also
in our environment we have G711 configured to a tandberg trunk so that
no matter what site your at desktop video calls to our tandberg
environment are G711 audio with a specific amount of video quality.

If the gateway and phones are on the same LAN then use G711 have them in
the same region. If the call is from a gateway to another site phone on
a different LAN over a WAN then use G729 between the regions.

My example for this would be a site gateway receives an ISDN call for a
phone, If the call is answered at this point it would be G711, but the
user is not available so the call diverts to voicemail, the voicemail
server is centrallised at the head office so the call traverses the WAN
to conenct to the voicemail box, this is now between regions so the call
is now G729. The gateway dial peer would need to support G729, the
voicemail server would need to support G729. If they don't both support
G729 then a transcoder is needed your preference here to reduce
bandwidth needed would be to have the gateway support G729 via the dial
peer and then a trancoder at head office next to the voicemail server so
that the traffic over the WAN from the gateway is G729 to the transcoder
and G711 to the G711 only voicemail server.

On Fri, Dec 11, 2009 at 1:49 PM, Aaron Riemer <ariemer at wesenergy.com.au>
wrote:

Thanks Daniel. So to make sure I understand this correctly. Can Cisco IP
phones dynamically use different codecs based on where calls are going
or coming from? If calls are coming from the PSTN then would it make
sense that the phone uses G.711 to reduce the level of transcoding
required? If calls are coming from the IP network then G.729 is used?
Can the call agent make this distinction of codec choice?

As you can no doubt probably guess I am very new to the area of voice
and it is difficult to know where to start!

Thanks,  

Aaron.

From: admin at danofive.id.au [mailto:admin at danofive.id.au] On Behalf Of
Daniel
Sent: Friday, 11 December 2009 11:32 AM
To: Aaron Riemer
Cc: cisco-voip at puck.nether.net 

Subject: Re: [cisco-voip] When is Transcoding Required?

Hi Aaron,

If you have a site with an ISDN 10 with a LAN that has phones configured
in the same region as the gateway set to use G729, and the gateway voice
dial peers (H323) set to use G729 then you won't need transcoding as
such. You will still need DSP's for the packetisation from ISDN to IP
packets. If you use G729 then you'll require a PVDM2-32 as you'll use
twice the amount because of the codec complexity. If you used G711 from
the gateway to phones then you'll only need a PVDM2-16.

To answer your question if a call comes in from the gateway requesting
only G711 to a device that is only G729 then yes a transcoder is
required preferably at the same LAN as the gateway and phone. For
instance if a site gateway recevies a call for a G711 UCCX call and the
UCCX server is in another region that requires the gateway to use G729
to UCCX then a transcoder is required to transcode the G729 call to G711
so that the UCCX prompts can be heard.

This would be the same for software conferencing from a branch site to
the core server, the branch phone and gateway would talk G729 to a
transcoder which would talk G711 to the software conference bridge. In
this situation a hardware conference bridge would be better utilised.

Use this link below about number of calls per DSP etc... High Complexity
codec would mean G729 to name one.

http://www.cisco.com/en/US/prod/collateral/modules/ps3115/ps6024/prod_qa
s0900aecd8016c6ad_ps3115_Products_Q_and_A_Item.html

hope that helps.

cheers,

Daniel

On Fri, Dec 11, 2009 at 12:36 PM, Aaron Riemer
<ariemer at wesenergy.com.au> wrote:

Hi Guys,

Can someone clarify exactly when transcoding between codec's will
actually occur on voice gateways? For example if I have a branch site
with Cisco IP Telephony and the phones are using G.729 and the site has
a voice gateway and PSTN services in what situation will transcoding
occur? Will all inbound and outbound calls through the PSTN require
transcoding? i.e. G.729 to G.711 and vice versa? If the site has 10
lines will I require at least 10 transcoding sessions to cater for 10
simultaneous PSTN calls?

Thanks,

Aaron.

LEGAL DISCLAIMER: This message contains confidential information and is
intended only for the individual named. If you are not the named
addressee you should not disseminate, distribute or copy this e-mail.
Please notify the sender immediately by e-mail if you have received this
e-mail by mistake and delete this e-mail from your system. If you are
not the intended recipient you are notified that disclosing, copying,
distributing or taking any action in reliance on the contents of this
information is strictly prohibited.

_______________________________________________
cisco-voip mailing list
cisco-voip at puck.nether.net
https://puck.nether.net/mailman/listinfo/cisco-voip

LEGAL DISCLAIMER: This message contains confidential information and is
intended only for the individual named. If you are not the named
addressee you should not disseminate, distribute or copy this e-mail.
Please notify the sender immediately by e-mail if you have received this
e-mail by mistake and delete this e-mail from your system. If you are
not the intended recipient you are notified that disclosing, copying,
distributing or taking any action in reliance on the contents of this
information is strictly prohibited.

LEGAL DISCLAIMER: This message contains confidential information and is intended only for the individual named. If you are not the named addressee you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately by e-mail if you have received this e-mail by mistake and delete this e-mail from your system. If you are not the intended recipient you are notified that disclosing, copying, distributing or taking any action in reliance on the contents of this information is strictly prohibited.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <https://puck.nether.net/pipermail/cisco-voip/attachments/20091211/9321c4b5/attachment.html>