The B Channel encodes voice using 8 8khz samples, which is 64kb.<br>
<br>
Once the B Channel is terminated at the router, the voice signal is
packetized into G711 using your DSPs, and an IP header is attached,
bringing the total bandwidth for the call to about 80kb.<br>
<br>
<br>
Jonathan<br><br><div><span class="gmail_quote">On 3/13/06, <b class="gmail_sendername">Kevin Thorngren</b> <<a href="mailto:kthorngr@cisco.com">kthorngr@cisco.com</a>> wrote:</span><blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">
AFAIK G.711 is 64 kbps. Take a look here for more info:<br><a href="http://www.cisco.com/en/US/tech/tk1077/">http://www.cisco.com/en/US/tech/tk1077/</a><br>technologies_tech_note09186a00800b6710.shtml<br><br>I would take a look at your firewall first. I don't remember seeing
<br>this in your attached drawing. The firewall might not be inspecting<br>the signaling packets properly for the audio to be setup correctly.<br>The CCM SRND has some basic info about firewalls.<br><a href="http://www.cisco.com/en/US/products/sw/voicesw/ps556/">
http://www.cisco.com/en/US/products/sw/voicesw/ps556/</a><br>products_implementation_design_guide_chapter09186a00805fba3a.html#wp1047<br>781<br><br>Kevin<br><br><br>On Mar 11, 2006, at 7:45 PM, new look wrote:<br><br>> Kevin,
<br>> <br>> <br>> Welcome back and thanks for your reply,<br>> <br>> 1-I want to ask you about the G711alaw I think it is need 66 kbps but<br>> as you know that the Voice channel in E1 is 64 Kbps only so is this
<br>> difference makes any problem? If I change the G711alaw to another<br>> Codec like G729 is this change solves the problem?<br>> <br>> 2-I have Firewall and put the CCM at server zone and the IP Phone and
<br>> the VGW in other zone.<br>> Is there is any problem in this design? Is it preferred to put the CCM<br>> and the VGW and the IP Phone at the same Subnet (zone)?<br>> <br>> Thanks<br>> Waleed<br>>
<br>> Kevin Thorngren <<a href="mailto:kthorngr@cisco.com">kthorngr@cisco.com</a>> wrote:<br>>> Sorry for the delay, I was busy the past couple of days and did not<br>>> have much time for email. You should reply to all that way someone
<br>>> else on the email list might respond and answer your remaining<br>>> questions.<br>>><br>>> G711alaw is fine to use with E1. The serial interface config is<br>>> specific to your environment so I wouldn't be able to tell you if
<br>>> something is needed or not. If it is working I would leave it as is.<br>>> I don't believe there is anything there that would cause one way audio<br>>> issues.<br>>><br>>> Kevin<br>>> On Mar 10, 2006, at 3:23 PM, new look wrote:
<br>>><br>>> > Dear Kevin,<br>>> > I'm still waiting your reply Thanks.<br>>> > <br>>> > Kevin,<br>>> > Thanks a lot for your fast response,<br>>> > First, the GW and the IP Phone are in the same subnet.
<br>>> > Second, what about the serial interface config. And the codec type?<br>>> > Thanks<br>>> > <br>>> ><br>>> > Kevin Thorngren wrote:<br>>> >> The first place to look is to make sure that you don't have any IP
<br>>> >> connectivity issues during the one way audio. Although your diagram<br>>> >> shows a pretty simple network but the phones may be on a different<br>>> >> subnet from the GW. You can hit the "i" or "?" button twice quickly
<br>>> >> while on the call. This will show you the call stats along the<br>>> RxCnt.<br>>> >> If the RxCnt is increasing then the RTP stream is getting to the<br>>> phone<br>>> >> but there is apparently blank audio if you are not hearing
<br>>> anything.<br>>> >> If the counter is not increasing then the problem is some sort of<br>>> IP<br>>> >> connectivity issue between the GW and the Phone.<br>>> >><br>>> >> You can use the "show voice dsp" command to see if there are any
<br>>> DSPs<br>>> >> that are having transmit or receive issues. Just look in the<br>>> >> TX/RX-PAK-CNT columns.<br>>> >><br>>> >> It doesn't look like dial peers 2,3,4 or 102 are needed. There are
<br>>> no<br>>> >> voice ports assigned to 2,3 or 4 and no session target assigned to<br>>> >> 102.<br>>> >><br>>> >> Kevin<br>>> >> On Mar 8, 2006, at 6:37 PM, new look wrote:
<br>>> >><br>>> >> > Dear All,<br>>> >> > I have a performance problemin my IP Phone system with the E1<br>>> calls<br>>> >> > (Some times I can't hear the person I called him), my system
<br>>> already<br>>> >> > configured before. (My network Diagram is attached)<br>>> >> > So first can any one help me to know if the existing<br>>> configuration<br>>> >> has
<br>>> >> > any wrong or unneeded commands? (The configuration of the VGWis<br>>> >> > attached)<br>>> >> > Note: My IP Phone DN#'s are 11XX<br>>> >> > <br>>> >> >
<br>>> >> > 1-I want to know if the interface Serial0/0/0:15 configurations<br>>> have<br>>> >> > any wrong commands or unneeded commands?<br>>> >> > 2-I want to know if I need dial-peers 2, 3 and 4, or not?
<br>>> >> > 3-I want to know if I need dial-peers 102, why?<br>>> >> > 4-Is the Codec g711ulaw is right codec for E1 calls or I need to<br>>> use<br>>> >> > another Codec? which one?
<br>>> >> > 5-is any one has the URL for Cisco example for this solution ?<br>>> >> ><br>>> >> > Yahoo! Mail<br>>> >> > Bring photos to life! New PhotoMail makes sharing a
breeze.isdn<br>>> >> > switch-type primary-net5<br>>> >> > !<br>>> >> > voice-card 0<br>>> >> > no dspfarm<br>>> >> > !<br>>> >> > voice translation-rule 1
<br>>> >> > rule 1 /^84/ /11/<br>>> >> > !<br>>> >> > voice translation-profile incom<br>>> >> > translate called 1<br>>> >> > !<br>>> >> > controller E1 0/0/0
<br>>> >> > framing NO-CRC4<br>>> >> > pri-group timeslots 1-31<br>>> >> > !<br>>> >> > translation-rule 1<br>>> >> > !<br>>> >> > !
<br>>> >> > interface Serial0/0/0:15<br>>> >> > no ip address<br>>> >> > isdn switch-type primary-net5<br>>> >> > isdn overlap-receiving<br>>> >> > isdn not-end-to-end 64
<br>>> >> > isdn incoming-voice voice<br>>> >> > isdn guard-timer 20000 on-expiry accept<br>>> >> > isdn map address * plan unknown type unknown<br>>> >> > isdn map address .* plan unknown type unknown
<br>>> >> > isdn map address transparent<br>>> >> > isdn T309-enable<br>>> >> > isdn T306 400000<br>>> >> > isdn sending-complete<br>>> >> > isdn incoming alerting add-PI
<br>>> >> > no isdn gtd<br>>> >> > no cdp enable<br>>> >> > !<br>>> >> > voice-port 0/0/0:15<br>>> >> > !<br>>> >> > dial-peer voice 1 pots
<br>>> >> > translation-profile incoming incom<br>>> >> > destination-pattern .T<br>>> >> > direct-inward-dial<br>>> >> > port 0/0/0:15<br>>> >> > !
<br>>> >> > dial-peer voice 2 pots<br>>> >> > preference 1<br>>> >> > destination-pattern .T<br>>> >> > !<br>>> >> > dial-peer voice 3 pots<br>>> >> > preference 2
<br>>> >> > destination-pattern .T<br>>> >> > !<br>>> >> > dial-peer voice 4 pots<br>>> >> > preference 3<br>>> >> > destination-pattern .T<br>>> >> > !
<br>>> >> > dial-peer voice 100 voip<br>>> >> > destination-pattern 1...<br>>> >> > session target ipv4:<a href="http://172.16.80.24">172.16.80.24</a><br>>> >> > dtmf-relay h245-alphanumeric
<br>>> >> > codec g711alaw<br>>> >> > !<br>>> >> > dial-peer voice 102 voip<br>>> >> > dtmf-relay h245-alphanumeric<br>>> >> > codec g711ulaw Yahoo! Mail
<br>>> > Use Photomail to share photos without annoying attachments.<br>> Yahoo! Mail<br>> Use Photomail to share photos without annoying attachments.<br><br>_______________________________________________
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</a><br><br><br></blockquote></div><br>