[cisco-voip] extension mobility
gokhan senol
gokhanciscottl at yahoo.com
Sat Nov 11 05:11:27 EST 2006
it you configure CCM extension mob. as autologout, yes when you login to 2nd phone the first phone will go autlogout.and the first phone will back to its normal directory number.
----- Original Message ----
From: "cisco-voip-request at puck.nether.net" <cisco-voip-request at puck.nether.net>
To: cisco-voip at puck.nether.net
Sent: Friday, November 10, 2006 11:43:30 PM
Subject: cisco-voip Digest, Vol 45, Issue 61
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Today's Topics:
1. Re: Optimizing RTP on the WAN. (Joe Pollere (US))
2. Re: auto QoS for Ip telephony solution (Ed Leatherman)
3. Extension Mobility (Cleaveland, AJ Allan @ IS)
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Message: 1
Date: Fri, 10 Nov 2006 16:13:35 -0500
From: "Joe Pollere \(US\)" <Joe.Pollere at us.didata.com>
Subject: Re: [cisco-voip] Optimizing RTP on the WAN.
To: "Joe Pollere \(US\)" <Joe.Pollere at us.didata.com>, "Scott ODonnell"
<scott.odonnell at gmail.com>, <cisco-voip at puck.nether.net>
Message-ID:
<C65033A817B5734392E55CB9DF44610403D8A23B at USNAEXCH.na.didata.local>
Content-Type: text/plain; charset="utf-8"
Sorry that should read "will be exaggerated beause of the larger payload size."
________________________________
From: cisco-voip-bounces at puck.nether.net on behalf of Joe Pollere (US)
Sent: Fri 11/10/2006 4:07 PM
To: Scott ODonnell; cisco-voip at puck.nether.net
Subject: Re: [cisco-voip] Optimizing RTP on the WAN.
You could change the default sampling rate from 20 ms to 30 ms but the bandwidth savings is really not that significant. However If you are experiencing any dropped packets it will be exasperated beause of the larger payload size.
See this for reference:
http://www.informit.com/articles/article.asp?p=357102&seqNum=1&rl=1
<snip>
Table 2-1 details the bandwidth per VoIP flow (both G.711 and G.729) at a default packetization rate of 50 packets per second (pps) and at a custom packetization rate of 33 pps. This does not include Layer 2 overhead and does not take into account any possible compression schemes, such as Compressed Real-Time Transport Protocol (cRTP, discussed in detail in Chapter 7, "Link-Specific Tools").
For example, assume a G.711 VoIP codec at the default packetization rate (50 pps). A new VoIP packet is generated every 20 ms (1 second / 50 pps). The payload of each VoIP packet is 160 bytes; with the IP, UDP, and RTP headers (20 + 8 + 12 bytes, respectively) included, this packet become 200 bytes in length. Converting bits to bytes requires multiplying by 8 and yields 1600 bps per packet. When multiplied by the total number of packets per second (50 pps), this arrives at the Layer 3 bandwidth requirement for uncompressed G.711 VoIP: 80 kbps. This example calculation corresponds to the first row of Table 2-1.
Table 2-1 Voice Bandwidth (Without Layer 2 Overhead)
Bandwidth Consumption
Packetization Interval
Voice Payload in Bytes
Packets Per Second
Bandwidth Per Conversation
G.711
20 ms
160
50
80 kbps
G.711
30 ms
240
33
74 kbps
G.729A
20 ms
20
50
24 kbps
G.729A
30 ms
30
33
19 kbps
NOTE
The Service Parameters menu in Cisco CallManager Administration can be used to adjust the packet rate. It is possible to configure the sampling rate above 30 ms, but this usually results in poor voice quality.
<snip>
HTH's
Joe
________________________________
From: cisco-voip-bounces at puck.nether.net on behalf of Scott ODonnell
Sent: Fri 11/10/2006 3:55 PM
To: cisco-voip at puck.nether.net
Subject: [cisco-voip] Optimizing RTP on the WAN.
I have a customer that is getting clobbered high WAN utilitization due to RTP streams.
While I recognize that at some point you just need more bandwidth are there any other ways to reduce the bandwidth RTP requires beyond codec selection and CRTP?
I vaguely remember back in the old "toll bypass" days, you could adjust the number of samples per packet or the sample rate and it could make a real difference in the bandwidth.
Any knobs like that in CallManager?
Scott
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Message: 2
Date: Fri, 10 Nov 2006 16:40:43 -0500
From: "Ed Leatherman" <ealeatherman at gmail.com>
Subject: Re: [cisco-voip] auto QoS for Ip telephony solution
To: "Erik Erasmus (E)" <ErasmuE4 at telkom.co.za>
Cc: cisco-voip at puck.nether.net
Message-ID:
<94a1afde0611101340r7729ed9ewf42f2e4f874b1118 at mail.gmail.com>
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Revisiting this thread since it seemed most related to my question.
As below, i've mainly been using "auto qos voip trust" on our interfaces
between cat 3750 switches. I've been going through the qos documentation,
particularly the SRND for enterprise qos - basically what it seems to be is
there are alot of differences between running "auto qos" and the recommended
configuration that they provide in the srnd.
One particular thing that just jumped out at me today was, using auto qos
voip trust on an interface doesn't appear to enable the priority queue
(according to the "sh mls qos int queueing" command). If that's true i've
got a ton of switches out there that need their configs fixed.. All my
traffic comes out marked OK, and queueing and so forth are setup OK on our
core routers but 90% of our distribution and access layer switches are 3550
or 3750 models.
So is auto-qos flat out not intended for anything but trusted endpoints such
as ccm servers? the further I dig into this the more it seems like this is
the case.
On 9/18/06, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:
>
> Thanks Ed
>
> I think it sound like good advice - I was basically planning to try it
> like you say.
> If one does not have extensive resources it is a lot easier and possibly
> good enough in most cases to use auto QoS.
>
> thanks for your advice - will let the group know if during my
> implementation I see something interseting.
>
> ------------------------------
> *From:* Ed Leatherman [mailto:ealeatherman at gmail.com]
> *Sent:* Mon 2006-09-18 16:29
> *To:* Erik Erasmus (E)
> *Cc:* cisco-voip at puck.nether.net
> *Subject:* Re: [cisco-voip] auto QoS for Ip telephony solution
>
> Hi Erik,
>
> I'm no qos expert by any stretch, this is just what we do and it seems to
> work out for us.. traffic markings come out end-to-end how we want them to
> at least. If someone else sees something wrong with it i'd love to hear it.
>
> I've been using "auto qos voip trust" on both those instances, i've been
> adding to that "mls qos trust dscp" to the callmanager ports and layer 3
> links, because the auto trust statement seems to put mls qos trust cos
> instead. To my understanding the callmanager servers mark their traffic with
> the dscp bits.
>
> Ed
>
> On 9/18/06, Erik Erasmus (E) <ErasmuE4 at telkom.co.za> wrote:
> >
> >
> > Currently at branches we use auto-QoS VoIP cisco-phone on all the 3560
> > series switchports connection to cisco phones and the same at HQ on the 4500
> > series switches.
> >
> >
> >
> > 1. I want to know what is recommended for the trunk interface on
> > the switch between the branch gateway and the lan switch on the
> > 802.1Q trunk and also on the router main / subinterfaces.
> > 2. Also what is recommended on switch ports connecting to the
> > Cisco Call Manager at HQ. Is it auto QoS VoIP trust ???
> >
> >
> Ed Leatherman
Senior Voice Engineer
West Virginia University
Telecommunications and Network Operations
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Message: 3
Date: Fri, 10 Nov 2006 15:08:45 -0600
From: "Cleaveland, AJ Allan @ IS" <Allan.J.Cleaveland at L-3Com.com>
Subject: [cisco-voip] Extension Mobility
To: <cisco-voip at puck.nether.net>
Message-ID:
<7AA7308B49C9A648929E81CD96CF18BC2A6D99 at gvlexch01.is.l-3com.com>
Content-Type: text/plain; charset="us-ascii"
A couple questions about Extension Mobility. I haven't used it before
but it's being looked at.
1) If I log into a phone go to another room and log into another phone
with this feature am I automatically logged out of the first phone?
2) Can I handle the logging in programmatically? Such as through an
API, adjust who's logged in to what and so forth?
Thank you,
Allan
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