[OSL | CCIE_Voice] Vol2 Lab 6 Question 4.2 Hardcode UCM Trunk Port
Hi, How can i hardcode the RAS port that used by the UCM trunk to register with GK to the port number that i want instead of the dynamically choose of by the system? I know you may put in the device name (gk-trunk) under UCM service param,but that's for port 1719 and 1720. Any clue? Shingei ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] How to test AAR?
Never mind. The bandwidth was configured on CUCM instead of the WAN interface. On Tue, Jan 4, 2011 at 11:50 AM, Michael Luo hout...@gmail.com wrote: I was doing ipexpert's CCIE Voice Volume 1 Lab 6A Task 6.1 - regarding Automated Alternate Route (AAR). In the Verification section, it says Test AAR by reducing the RSVP BANDWIDTH allocated to either the HQ or BR1 WAN interface down to 39kbps, and then making a call from HQ Phone 2 to BR1 Phone 2. How exactly do I do that? I applied the command ip rsvp bandwidth 39 to HQ/BR1 serial (frame-relay) interface, but the call still goes through IP network instead of PSTN. How does CallManager know the bandwidth has been decreased? Thanks! Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCM Huntgroup (No Fwd)
Hi Gents, I'm working on Hunt group configuration on CUCM. I wanted the CallFwdNoAn and CallFwdBusy settings shoud be honored for the user who forwarded the call instead of the Hunt pilot/ Below My config: One Hunt Group One Hunt List One Hunt Pilot (5000) and I've checked Use Personal preferences for CallFwdNoAn and CallFwdBusy The problem is whenever I receive a call on the hunt Pilot the same ring in circular fashion but the CallFwdNoAn and CallFwdBusy doesn't happened even the forward settings has been set on the members of the hunt group. Simply I heard busy tone without forward. Any Ideas? Regards, Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UC user login
Hi harooon, Thank you for your help. Unfortunately, RTMT returns this message no files matches the date range for node ... Is required tracing activated by default on UC to collect these files? Thanks Julien Le 4 janv. 2011 à 11:21, haroon javed harooon.ja...@gmail.com a écrit : dear Julien, Use RTMT -- trace and log center-- collect file --- Role base security. and use MLA on Unity Connection. Regards Haroon On Tue, Jan 4, 2011 at 1:48 PM, Julien Krieger julien.krie...@ineo-gdfsuez.com wrote: Hi all, I have a unity connection without LDAP integration. I would like to monitor users who try to log in. Is there any tools or ways available to monitor user login attempt/failed? Julien ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, Haroon Javed Telecom Engineer Cell: +92 (321) 8430260 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Access
Hi Roger, MVA on a MGCP control gateway is possible.In fact,that is a coexisting of both MGCP and H323 on the same gateway,but you could not used a MGCP control PRI for MVA. you may refer to Netpro for own interest. https://supportforums.cisco.com/thread/2005673 Shingei. On Sun, Jan 2, 2011 at 11:16 PM, Shrini linuxbos...@gmail.com wrote: Hi ShinGei , bkvalentine, Rogers et al I remember it was successful last time when I configured it another lab when HQ was h323. Now I was confused around dial-peers hence had the question. I will give a try now with MGCP + H323 on HQ and it should work. Thanks all. Shrini *From:* ShinGei Yong [mailto:shingei.y...@gmail.com] *Sent:* Sunday, January 02, 2011 6:42 AM *To:* Shrini; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Mobile Voice Access Hi Shrini, What about your UCM configuration? 1. is your H323 GW registered with UCM? 2. what is your dialing behavior internally?4 or 10?if is 4, then your in outbound dp should be 4 digit patten as well instead of 10. Please provide more info Shingei On Sun, Jan 2, 2011 at 4:13 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, *Wish you all a Happy and Prosperous New Year 2011* First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] UC user login
Hi all, I have a unity connection without LDAP integration. I would like to monitor users who try to log in. Is there any tools or ways available to monitor user login attempt/failed? Julien ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] UC user login
dear Julien, Use RTMT -- trace and log center-- collect file --- Role base security. and use MLA on Unity Connection. Regards Haroon On Tue, Jan 4, 2011 at 1:48 PM, Julien Krieger julien.krie...@ineo-gdfsuez.com wrote: Hi all, I have a unity connection without LDAP integration. I would like to monitor users who try to log in. Is there any tools or ways available to monitor user login attempt/failed? Julien ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com -- Regards, *Haroon Javed* *Telecom Engineer* Cell: +92 (321) 8430260 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CTI Route Point - Forwarding and Caller ID question
The reason is that each DID is associated with a user's fax, which will end up in their mailbox. It uniquely identifies them. My idea was to translate it into another unique number, and then pass those digits to the hunt group and into the fax. But another colleague reminded me that SCCP controlled devices are unable to pass those sorts of digits. I've been recommended to instead use a voice gateway, controlled by a protocol such as h.323 or SIP trunk. And instead of using translation patterns, using route patterns that are pointed to the route list for the proper gateway. Thanks, Mark On Tue, Jan 4, 2011 at 1:40 PM, basant yadav basant.ya...@gmail.com wrote: Mark Not sure if it'll help but when you are translating the DID to 4 digit on CUCM, why don't you translate it directly to 4321 (HG number) as oppose to translating it to 1234 first and then fwding it to HG number 4321? My 2 Cents.. - Basant On Tue, Jan 4, 2011 at 5:16 PM, Mark Davis davismar...@gmail.comwrote: How do I pass a called number through an ATA 187 analog connection? This is in regards to the fax server on an old PBX that is currently connected using two analog ports. There are multiple DIDs that translate to different extensions that then get forwarded to a hunt group with the two analog connections as members. It’s important that the translated extensions get passed to the fax server because that is how the fax server knows how to forward the fax to the right person. Here's my idea/scenario: 10 digit DID is translated within UCM to a 4 digit extension, such as 1234. I have a CTI route point with the extension 1234 that I want to forward to the hunt group 4321. Hunt group 4321 has two members: 4567, 9876. Both 4567 and 9876 belong to a single ATA with two analog lines connected to a fax server. I need the number 1234 sent thru to and received by the fax server. Would this be accomplished by using the Forwarded Call Information Display on the CTI route point and selecting Redirected Number? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Book 1 lab 5.5
I'm reviewing the video walk thro lab5.5. In the lab they set localized call routing for hq and br1 with local partitions for pdt access. Why do they not use local call routing group for a least 11 digit dialing? 11 digit Set up a generic route pattern of \+91212.! and make use of lock router gap as per dp of the handset Or even also have a translation of 9.! With a discard credit and s prefix of 91212 and then point it to the generic route pattern \+91212.! 7digit I agree a local pstn per site is needed Set up a translation of [2-9]xx Assign local pstn partition and point it to the generic route pattern Thanks ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] PSTN simulation with h323
Hi all, I was wondering if there is anyone else struggling with PSTN configuration via h323. I have a lab built on dynamips, and am using one router as PSTN, and I have voip dial-peers towards this router. To be able to support this, I have all routers (BR1, BR2, HQ) configured as CUBE, to pass calls ip to ip, which involves many problems in this kind of configuration. Here are a few problems that I have encountered. 1) CUCM --- h323 --- CUBE --- h323 --- PSTN In this case, we have two options: - either enable fast-start - or remove Wait for Far End H.245 Terminal Capability Set on gateway configuration on a Call Manager and then it will work without MTP too. 2) CUCM --- SIP --- CUBE --- h323 --- PSTN We have one option here: - MTP again, to allow early media on SIP In the second scenario, when the call comes in from PSTN, it works. As IOS Gateways are configured for h323 fast-start, and SIP is configured for early media, therefore it works fine. but the problem is on calls from CUCM towards PSTN because without MTP the call is going out with SIP media delay therefore without SDP, and h245 negotiation on h323 to PSTN never happens. Does anyone have any idea what can be done to make it work without MTP??? Thanks all for help, George, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] QoS suggestion
Hi All , I have just started the QoS preparation and gone through books and SRND. I don’t want to go through entire material and just want to concentrate on the hot topic which is asked frequently . From the forum and friends ,i concluded that MLP and FRF.12 is asked frequently . Want to discuss and need suggestion if such question is asked For FRF.12 , I guess , I need to copy the cli from SNRD Following cli will be copied policy-map MQC-FRTS-768 class class-default shape average 729600 7296 0 ! Enables MQC-Based FRTS service-policy WAN-EDGE ! Queues packets headed to the shaper ! … ! interface Serial2/0 no ip address encapsulation frame-relay ! interface Serial2/0.12 point-to-point ip address 10.1.121.1 255.255.255.252 description 768kbps FR Circuit to RBR-3745-Left frame-relay interface-dlci 102 class FR-MAP-CLASS-768 ! Binds the map-class to the FR DLCI ! … ! map-class frame-relay FR-MAP-CLASS-768 service-policy output MQC-FRTS-768 ! Attaches nested MQC policies to map-class frame-relay fragment 960 ! Enables FRF.12 class-map match-all Voice match ip dscp ef ! IP Phones mark Voice to EF class-map match-any Call Signaling match ip dscp cs3 ! Future Call-Signaling marking match ip dscp af31 ! IP Phones mark Call-Signaling to AF31 ! policy-map WAN-EDGE class Voice priority percent 33 ! Maximum recommended LLQ value compress header ip rtp ! Optional: Enables Class-Based cRTP class Call Signaling bandwidth percent 5 ! BW guarantee for Call-Signaling class class-default fair-queue ! All other data gets fair-queuing ! class-map match-all BEST-EFFORT match any or even: class-map match-all BEST-EFFORT match access-group 101 ... access-list 101 permit ip any any *Based on the bandwidth , I can modify the fragmentation size* . For MLP question , I guess I just need to configure “auto qos voip trust fr-atm” under specific PVC Here I have one doubt , when auto qos voip trust fr-atm is configured , it means we are filtering packets on the marked DSCP value, in that case , do I need to create access list to filter based on the IP address/ protocol or QoS should be configured on switch also to mark the packet based on ip address ? Please suggest ! Thx, Rahul ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CTI Route Point - Forwarding and Caller ID question
Mark Not sure if it'll help but when you are translating the DID to 4 digit on CUCM, why don't you translate it directly to 4321 (HG number) as oppose to translating it to 1234 first and then fwding it to HG number 4321? My 2 Cents.. - Basant On Tue, Jan 4, 2011 at 5:16 PM, Mark Davis davismar...@gmail.com wrote: How do I pass a called number through an ATA 187 analog connection? This is in regards to the fax server on an old PBX that is currently connected using two analog ports. There are multiple DIDs that translate to different extensions that then get forwarded to a hunt group with the two analog connections as members. It’s important that the translated extensions get passed to the fax server because that is how the fax server knows how to forward the fax to the right person. Here's my idea/scenario: 10 digit DID is translated within UCM to a 4 digit extension, such as 1234. I have a CTI route point with the extension 1234 that I want to forward to the hunt group 4321. Hunt group 4321 has two members: 4567, 9876. Both 4567 and 9876 belong to a single ATA with two analog lines connected to a fax server. I need the number 1234 sent thru to and received by the fax server. Would this be accomplished by using the Forwarded Call Information Display on the CTI route point and selecting Redirected Number? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Access
Hi Jeff, Thanks for your valuable suggestion. It worked. -S _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of givemeccievoice2...@gmail.com Sent: Sunday, January 02, 2011 2:50 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hi Shrini, If you follow the Features and Services Guide as mentioned before, you will have success. You need to configure hairpinning for MGCP to work with MVA. The idea is that you will accept the call using 5999, but the MVA pilot number will be a different number. You will have to add the h323 gateway to CUCM and create a route pattern that will direct calls for 5999 to the h323 gateway. The incoming dialpeer on the gateway will be the 5999 number which will trigger the VXML script. There will be a different number needed for the MVA pilot, for example 6000. The outgoing dial-peer will point back to CUCM using this number (6000). As the h323 gateway and the MGCP gateway will be logically separate (listening to different interfaces), you can accomplish this on the same box. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Carpio Sent: Sunday, January 02, 2011 7:40 AM To: ShinGei Yong Cc: ccie_voice@onlinestudylist.com; Shrini Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hello ShinGei, Thanks for the info. In the end, MGCP will not make it for us Regards, Roger Carpio. On Sun, Jan 2, 2011 at 9:34 AM, ShinGei Yong shingei.y...@gmail.com wrote: Hi Roger, MVA on a MGCP control gateway is possible.In fact,that is a coexisting of both MGCP and H323 on the same gateway,but you could not used a MGCP control PRI for MVA. you may refer to Netpro for own interest. https://supportforums.cisco.com/thread/2005673 Shingei. On Sun, Jan 2, 2011 at 11:16 PM, Shrini linuxbos...@gmail.com wrote: Hi ShinGei , bkvalentine, Rogers et al I remember it was successful last time when I configured it another lab when HQ was h323. Now I was confused around dial-peers hence had the question. I will give a try now with MGCP + H323 on HQ and it should work. Thanks all. Shrini From: ShinGei Yong [mailto:shingei.y...@gmail.com] Sent: Sunday, January 02, 2011 6:42 AM To: Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hi Shrini, What about your UCM configuration? 1. is your H323 GW registered with UCM? 2. what is your dialing behavior internally?4 or 10?if is 4, then your in outbound dp should be 4 digit patten as well instead of 10. Please provide more info Shingei On Sun, Jan 2, 2011 at 4:13 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all a Happy and Prosperous New Year 2011 First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Access
Hi Shrini, If you follow the Features and Services Guide as mentioned before, you will have success. You need to configure hairpinning for MGCP to work with MVA. The idea is that you will accept the call using 5999, but the MVA pilot number will be a different number. You will have to add the h323 gateway to CUCM and create a route pattern that will direct calls for 5999 to the h323 gateway. The incoming dialpeer on the gateway will be the 5999 number which will trigger the VXML script. There will be a different number needed for the MVA pilot, for example 6000. The outgoing dial-peer will point back to CUCM using this number (6000). As the h323 gateway and the MGCP gateway will be logically separate (listening to different interfaces), you can accomplish this on the same box. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roger Carpio Sent: Sunday, January 02, 2011 7:40 AM To: ShinGei Yong Cc: ccie_voice@onlinestudylist.com; Shrini Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hello ShinGei, Thanks for the info. In the end, MGCP will not make it for us Regards, Roger Carpio. On Sun, Jan 2, 2011 at 9:34 AM, ShinGei Yong shingei.y...@gmail.com wrote: Hi Roger, MVA on a MGCP control gateway is possible.In fact,that is a coexisting of both MGCP and H323 on the same gateway,but you could not used a MGCP control PRI for MVA. you may refer to Netpro for own interest. https://supportforums.cisco.com/thread/2005673 Shingei. On Sun, Jan 2, 2011 at 11:16 PM, Shrini linuxbos...@gmail.com wrote: Hi ShinGei , bkvalentine, Rogers et al I remember it was successful last time when I configured it another lab when HQ was h323. Now I was confused around dial-peers hence had the question. I will give a try now with MGCP + H323 on HQ and it should work. Thanks all. Shrini From: ShinGei Yong [mailto:shingei.y...@gmail.com] Sent: Sunday, January 02, 2011 6:42 AM To: Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hi Shrini, What about your UCM configuration? 1. is your H323 GW registered with UCM? 2. what is your dialing behavior internally?4 or 10?if is 4, then your in outbound dp should be 4 digit patten as well instead of 10. Please provide more info Shingei On Sun, Jan 2, 2011 at 4:13 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all a Happy and Prosperous New Year 2011 First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com image001.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Voicemail - Time received
Hi, In Unity Connection Voicemail. When I listen the VM on HQ phone or BR1 Phone the time received in the envelope is always the time on Unity Connection, where are HQ and BR1 are in PST and EST respectively. How can I get local time in the message envelope. thanks Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Thanks! I will try these suggestions as well, From: Roger Källberg [mailto:roger.kallb...@cygate.se] Sent: Saturday, January 01, 2011 12:06 PM To: Hough, Earl; ccie_voice@onlinestudylist.com Subject: SV: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Hi Earl, You might want to add the bind interface statement under your ccm group. I remember that I had strange issues when I didn't have that line in my config back when I did my preparation for the lab. I'm not sure if that has to do with your specific issue, but it's worth a try :-) In your case it would be HQ sccp ccm group 1 bind interface FastEthernet0/0.20 BR2 sccp ccm group 1 bind interface Loopback0 And one note, that has absolutely nothing to do with this issue, but is a good thing to remember to always do. Remove all codecs with a b in the name from your transcoder and if also setup conference dspfarm profiles. Remember vad is bad Best of luck. Sincerely Roger Källberg CCIE #26199 (Voice) Consultant Cygate AB Eric Perssons väg 21, SE-217 62 MALMÖ Från: Hough, Earl [earl.ho...@pcmallservices.com] Skickat: den 31 december 2010 15:04 Till: ccie_voice@onlinestudylist.com Ämne: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Everyone, Been struggling with a scenario which seems to be rock-solid for SCCP endpoints. The topology I'm working with is as follows: BR2 (UCME 7.0) G.729r8-- HQ (GK/CUBE) --G.711ulaw-- BR2 (UCME 7.0) What I've been trying to do is to connect the two remote branches as H323 gateways utilizing the GK/CUBE resources of the HQ router to provide GK address resolution and media termination. I have no problems getting this to work using viazone processing in both directions when using exclusively SCCP endpoints. What I have a problem with is SIP endpoints on BR2 being able to call SCCP endpoints at BR1. The SIP endpoints at BR2 will initiate a call to a SCCP phone at BR1 and the BR1 phone will ring, but after the call is answered on the BR1 side, the BR2 side continues to ring out and never completes the call. If I hang up the call from the BR1 side both sides disconnect, so it appears as though signaling is still working, just not the media path. It also appears as though the H245 capability set is never completed when a SIP endpoint at BR2 initiates a call to BR1. It does correctly work when an SCCP endpoint at BR2 initiates a call to BR1. I've been scratching my head looking over debugs and traces for several hours here and though I'd throw it out to the group as to what anyone's thoughts would be as to why this isn't working correctly. If I go straight through from BR2 to BR1 only using GK address resolution and not via-zone processing in that direction, the SIP endpoints are able to complete calls. Any thoughts on this from group? The relevant config portions are as follows: HQ-RTR (GK/CUBE) --- ! voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip ! ! voice-card 0 no dspfarm dsp services dspfarm ! ! interface Loopback0 ip address 10.10.110.1 255.255.255.0 h323-gateway voip interface h323-gateway voip id CUBE ipaddr 10.10.110.1 1719 h323-gateway voip h323-id HQ-RTR h323-gateway voip bind srcaddr 10.10.110.1 ! ! sccp local FastEthernet0/0.20 sccp ccm 10.10.200.3 identifier 1 version 6.0 sccp ! sccp ccm group 1 associate ccm 1 priority 1 associate profile 1 register XCD-CME ! dspfarm profile 1 transcode codec g711ulaw codec g711alaw codec g729ar8 codec g729abr8 codec g729r8 maximum sessions 4 associate application SCCP ! ! dial-peer voice 3000 voip incoming called-number 3...$ dtmf-relay h245-alphanumeric codec g711ulaw no vad ! dial-peer voice 4000 voip destination-pattern 3...$ session target ras dtmf-relay h245-alphanumeric no vad ! dial-peer voice 3100 voip incoming called-number 1...$ dtmf-relay h245-alphanumeric no vad ! dial-peer voice 4100 voip destination-pattern 1...$ session target ras dtmf-relay h245-alphanumeric codec g711ulaw no vad ! ! gateway timer receive-rtp 1200 ! ! ! gatekeeper zone local BR1 cisco.com 10.10.110.1 outvia CUBE zone local CUBE cisco.com zone local BR2 cisco.com outvia CUBE zone prefix BR1 1... zone prefix BR2 3... gw-type-prefix 1#* default-technology no shutdown ! ! telephony-service sdspfarm units 1 sdspfarm transcode sessions 4 sdspfarm tag 1 XCD-CME load 7961 SCCP41.8-3-3S load 7962 SCCP42.8-3-3S load 7965 SCCP45.8-3-3S max-ephones 10 max-dn 20 no-reg both ip source-address 10.10.200.3 port 2000 max-conferences 8 gain -6 transfer-system full-consult create cnf-files version-stamp 7960 Dec 30 2010 07:46:37 ! BR1 (CME w/ SCCP-only endpoints) -
[OSL | CCIE_Voice] CUC Integration to CME and SIP Phones
Hello People, today i reviewed CUC and CUE integration with CME trying to analize different scenario.I succesfully test both CUC and CUE integration Using SIP towards CME, but i had some problem testing CUC integration using SCCP versus a CME. The integration guide report that it should be possible to integrate both SCCP and SIP Phones. On my side SCCP Phones works fine but i'm not able to light the mwi for SIP Phones. For SIP Phones i've configured the MWI server ( i've tryed both subscribe notify and unsolicited ) but when i make a call to a sip phone CUC accept the message and try to light the MWI but i've the following error: Jan 2 16:56:38.661: //-1//SIP/Msg/ccsipDisplayMsg:Sent:NOTIFY sip:3...@10.10.201.69:5060;transport=udp SIP/2.0Date: Sun, 02 Jan 2011 16:56:38 GMTFrom: sip:3...@10.10.112.2;tag=E96EE8-1525Event: message-summaryContent-Length: 23User-Agent: Cisco-SIPGateway/IOS-12.xTo: sip:3...@10.10.201.69Contact: sip:3...@10.10.202.1:5060Content-Type: application/message-summaryCall-ID: 19809ad5-15c811e0-8e088e4e-fc63f...@10.10.202.1subscription-state: activeVia: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bKC1C729CSeq: 101 NOTIFYMax-Forwards: 70 Messages-Waiting: yes BR2-RTR#Jan 2 16:56:38.865: //-1//SIP/Msg/ccsipDisplayMsg:Received:SIP/2.0 400 Bad RequestVia: SIP/2.0/UDP 10.10.202.1:5060;branch=z9hG4bKC1C729From: sip:3...@10.10.112.2;tag=E96EE8-1525To: sip:3...@10.10.201.69Call-ID: 19809ad5-15c811e0-8e088e4e-fc63f...@10.10.202.1date: Sun, 02 Jan 2011 16:56:34 GMTWarning: 399 Bad MWI NOTIFYCSeq: 101 NOTIFYContent-Length: 0 Any ideas? or this type of integration is supposed to work only for SCCP Phones? Regards Matteo ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Video walk thro. Book 1 lab 5
I'm viewing the video and have a question on route patterns. In the video and books they go through localized call routing 5.5. They set up local p ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM Huntgroup (No Fwd)
You must also set a call forward no coverage (and the associated CSS) on the dn that forwarded the call to the hunt pilot. HTH Sent from my Verizon Wireless Phone - Reply message - From: Naoufal Kerboute naou...@mhdinfotech.com Date: Sun, Jan 2, 2011 4:06 am Subject: [OSL | CCIE_Voice] CUCM Huntgroup (No Fwd) To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Hi Gents, I'm working on Hunt group configuration on CUCM. I wanted the CallFwdNoAn and CallFwdBusy settings shoud be honored for the user who forwarded the call instead of the Hunt pilot/ Below My config: One Hunt Group One Hunt List One Hunt Pilot (5000) and I've checked Use Personal preferences for CallFwdNoAn and CallFwdBusy The problem is whenever I receive a call on the hunt Pilot the same ring in circular fashion but the CallFwdNoAn and CallFwdBusy doesn't happened even the forward settings has been set on the members of the hunt group. Simply I heard busy tone without forward. Any Ideas? Regards, Naoufal * * This Communication is Private Confidential. This message and any attachments may contain information that is privileged and / or confidential and is the property of MHD InfoTech LLC. * * It is intended solely for the person to whom it is addressed. If you are not the intended recipient, you are hereby notified that you are not authorized to read, print, retain copy, disseminate, distribute, or * * use this message any attachments or any part thereof. If you have received this message in error, please notify the sender immediately and delete the message and any attachments from your system. * * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco Presence Issue
In addition: Check if you configured CUPS on CUCM under application server _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of shafqut.ha...@gmail.com Sent: Sunday, January 02, 2011 11:28 PM To: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Cisco Presence Issue To add one thing more that everything is green on System Troubleshooting page except the warnings for components which I haven't configured yet. Regards, Shamid Sent from my BlackBerryR smartphone using my Telenor Persona connection _ From: shafqut hamid shafqut.ha...@gmail.com Date: Mon, 3 Jan 2011 11:05:42 +0500 To: ccie_voice@onlinestudylist.com Subject: Cisco Presence Issue I have configured CUPS but when viewing topology page its showing red cross on server with following description * SIP Proxy (UNKNOWN) * Presence Engine (UNKNOWN) * Presence Engine Database (UNKNOWN) * CUP Database (UNKNOWN) * Sync Agent (UNKNOWN) * Inter-Cluster Sync Agent (UNKNOWN) while checking server status on CUPC its showing everything green except voicemail (Failed to Connect-No user Credential) and LDAP (Not Available-Invalid Credentials). Pictures are attached for your reference. Can someone please advise where to look to resolve these? Regards, Shamid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CTI Route Point - Forwarding and Caller ID question
How do I pass a called number through an ATA 187 analog connection? This is in regards to the fax server on an old PBX that is currently connected using two analog ports. There are multiple DIDs that translate to different extensions that then get forwarded to a hunt group with the two analog connections as members. It’s important that the translated extensions get passed to the fax server because that is how the fax server knows how to forward the fax to the right person. Here's my idea/scenario: 10 digit DID is translated within UCM to a 4 digit extension, such as 1234. I have a CTI route point with the extension 1234 that I want to forward to the hunt group 4321. Hunt group 4321 has two members: 4567, 9876. Both 4567 and 9876 belong to a single ATA with two analog lines connected to a fax server. I need the number 1234 sent thru to and received by the fax server. Would this be accomplished by using the Forwarded Call Information Display on the CTI route point and selecting Redirected Number? Thanks, Mark ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
After playing with codec settings on both sides of the GK/CUBE the results I am experiencing are as follows. In each of these tests I am able successfully call from a SCCP endpoint at BR2 to a SCCP endpoint at BR1. And in each of these tests, regardless of the WAN codec, the SIP endpoints are using G711ulaw as defined in voice register pool. BR2 (Sip Endpoint) - (G.711ulaw) -- HQ-RTR (GK/CUBE) -- (G.711ulaw) BR1*SUCCESSFUL* BR2 (Sip Endpoint) - (G729r8) -- HQ-RTR (GK/CUBE) -- (G.729r8) BR1*SUCCESSFUL* BR2 (Sip Endpoint) - (G729r8) -- HQ-RTR (GK/CUBE) -- (G.711ulaw) - BR1*NOT SUCCESSFUL* BR2 (Sip Endpoint) - (G711ulaw) --- HQ-RTR (GK/CUBE) -- (G.729r8) BR1*NOT SUCCESSFUL* So, it appears as though when both legs passing through the HQ-RTR GK/CUBE instance are the same, the calls from SIP endpoints can complete. Whenever the codecs are different, the calls cannot complete. From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Saturday, January 01, 2011 6:54 AM To: John Nield Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing I would agree with John here as Software MTP would not support G729. This might be the issue with your case. An MTP needs to be pre-allocated for early-offer calls, you must configure an external MTP or transcoder device to use this feature. The software MTP does not support G.729 over SIP trunks. Do a show sccp connection while making the call to BR2 and see if you can see an MTP invoked? Ash CCIE#26244 On 01/01/2011 10:27, John Nield wrote: On 1/01/2011 3:37 PM, ccie_voice-requ...@onlinestudylist.com wrote: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Hi sounds like you br1 xcoders is not being allocated to the call, therefore you have no MTP function and a world of hurt. this theory assumes you're HQ phones are ok. to verify this run up rtmt and check the number of xcoding resources being used, also check if the graph for unable to allocate resources. i assume your br1 MRLG contains a hardware xcoder that would be used for this instance, a software MTP in my experience fails due to the g729 codec the br1 sites will be requesting. good luck. regards john ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Access
Hi ShinGei , bkvalentine, Rogers et al I remember it was successful last time when I configured it another lab when HQ was h323. Now I was confused around dial-peers hence had the question. I will give a try now with MGCP + H323 on HQ and it should work. Thanks all. Shrini From: ShinGei Yong [mailto:shingei.y...@gmail.com] Sent: Sunday, January 02, 2011 6:42 AM To: Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Mobile Voice Access Hi Shrini, What about your UCM configuration? 1. is your H323 GW registered with UCM? 2. what is your dialing behavior internally?4 or 10?if is 4, then your in outbound dp should be 4 digit patten as well instead of 10. Please provide more info Shingei On Sun, Jan 2, 2011 at 4:13 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, Wish you all a Happy and Prosperous New Year 2011 First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Yeah, I was playing with the three options I see under “voice service voip - h323 - call start”. The results are pasted below. Also, from the Cisco Voice Gateways and Gatekeepers text, on page 58 of the H323 chapter: “You do not have to configure Fast Start – it is used by default unless any of the following conditions is present: · One of the gateways either does not support or rejects the use of Fast Start. The gateways signals this by the absence of a FastStart elemtn in any of the setup signals, up to and including the Connect message. · The called gateway selects different codecs for sending and receiving · Recent versions of the Cisco IOS use Fast Start for calls that RSVP initiates. You might have to enable slow start for backward compatibility with older versions. When any of these occur, separate H.245 channels are opened, and the sending and receiving capabilities are negotiated the slow way.” Plus, when I look at the output of “debug cch323 all” I can tell that FastStart procedures are being initiated. BR2-RTR(config)#voice service voip BR2-RTR(conf-voi-serv)#h323 BR2-RTR(conf-serv-h323)#call start fast BR2-RTR(conf-serv-h323)#do show run | section voice service voip voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 sip registrar server BR2-RTR(conf-serv-h323)#call start slow BR2-RTR(conf-serv-h323)#do show run | section voice service voip voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 call start slow sip registrar server BR2-RTR(conf-serv-h323)#call start interwork BR2-RTR(conf-serv-h323)#do show run | section voice service voip voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 call start interwork sip registrar server BR2-RTR(conf-serv-h323)#default call start BR2-RTR(conf-serv-h323)#do show run | section voice service voip voice service voip allow-connections h323 to h323 allow-connections h323 to sip allow-connections sip to h323 allow-connections sip to sip h323 sip registrar server BR2-RTR(conf-serv-h323)# From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Saturday, January 01, 2011 10:07 AM To: Hough, Earl Cc: Ashar Siddiqui; John Nield; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Sorry, I missed the part where you mention these are 2 CMEs. But regarding Fast Start - I've just checked that my router never shows this command in running config. Both call start fast and no call start fast don't show up. I can't figure out any other command to verify this at the moment, but I was convinced that Fast Start is not default option. It won't hurt to try it. Let us know. RS30-CCIE(config)#voice service voip RS30-CCIE(conf-voi-serv)#h323 RS30-CCIE(conf-serv-h323)#do sh run | s h323 h323 RS30-CCIE(conf-serv-h323)#call start fast RS30-CCIE(conf-serv-h323)#do sh run | s h323 h323 RS30-CCIE(conf-serv-h323)#no call start fast RS30-CCIE(conf-serv-h323)#do sh run | s h323 h323 RS30-CCIE(config)# On Sat, Jan 1, 2011 at 15:58, Hough, Earl earl.ho...@pcmallservices.com wrote: I agree you need to ensure that fast start is enabled in order to facilitate SIP Early Offer - H323 Fast Start interworking. It was always my understanding that H323 fast start was enabled by default on IOS platforms. In fact, when you try to enable the command “call start fast” under voice service voip - h323, the command doesn’t show up in the output of running-config. Also, this scenario doesn’t involve UCM. It was two separate CME sites. _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Mobile Voice Access
Hi Experts, Wish you all a Happy and Prosperous New Year 2011 First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Voicemail - Time received
hi, set proper timezone for your user in CUC. hth On Jan 2, 2011 12:02 PM, Shrini linuxbos...@gmail.com wrote: Hi, In Unity Connection Voicemail. When I listen the VM on HQ phone or BR1 Phone the time received in the envelope is always the time on Unity Connection, where are HQ and BR1 are in PST and EST respectively. How can I get local time in the message envelope. thanks Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Sorry, I don't fully understand your architecture (in your first post you indicated you want to have different codecs on both CUBE call legs). But if you want to do transcoding on CUBE you need to register transcoder on it (via telephony services). On Sat, Jan 1, 2011 at 15:36, Hough, Earl earl.ho...@pcmallservices.comwrote: After playing with codec settings on both sides of the GK/CUBE the results I am experiencing are as follows. In each of these tests I am able successfully call from a SCCP endpoint at BR2 to a SCCP endpoint at BR1. And in each of these tests, regardless of the WAN codec, the SIP endpoints are using G711ulaw as defined in “voice register pool”. BR2 (Sip Endpoint) - (G.711ulaw) -- HQ-RTR (GK/CUBE) -- (G.711ulaw) BR1**SUCCESSFUL** BR2 (Sip Endpoint) - (G729r8) -- HQ-RTR (GK/CUBE) -- (G.729r8) BR1**SUCCESSFUL** BR2 (Sip Endpoint) - (G729r8) -- HQ-RTR (GK/CUBE) -- (G.711ulaw) - BR1**NOT SUCCESSFUL** BR2 (Sip Endpoint) - (G711ulaw) --- HQ-RTR (GK/CUBE) -- (G.729r8) BR1**NOT SUCCESSFUL** So, it appears as though when both legs passing through the HQ-RTR GK/CUBE instance are the same, the calls from SIP endpoints can complete. Whenever the codecs are different, the calls cannot complete. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
+ enable Fast Start on CUBE and CUCM trunk (enable inbound fast start or sth similar). On Sat, Jan 1, 2011 at 15:49, Miron Kobelski findko...@gmail.com wrote: Hi, SIP phones by default originate outgoing calls using SIP Early Offfer. When the next hop is H.323 trunk, you might want to try enabling Fast Start on it (voice services voip or voice class h323 - call start fast). That might solve your issue with H245 negotiation. HTH kobel On Sat, Jan 1, 2011 at 15:19, Hough, Earl earl.ho...@pcmallservices.comwrote: I never had a software MTP defined. It was always a hardware transcoder at BR2 and of course, the one collocated with the GK at HQ. I’m not having an issue calling from BR1 to BR2. The issue is when I call from a SIP endpoint at BR2 back to BR1. To recap the scenario and hopefully clear up any confusion, the SIP endpoints at BR2 are using G711ulaw as defined in the voice register pool. There is a hardware transcoder at BR2 which supports the G729r8 codec. The leg between BR2 and HQ is using G729r8. The HQ GK/CUBE reterminates the media path and sends the call on to BR1. The leg between HQ and BR1 is G711ulaw. The phone at the BR1 site is an SCCP endpoint. When the calls at BR2 originate from an SCCP endpoint, the behavior is as expected and displayed below: HQ-RTR#show gatekeeper calls Total number of active calls = 2. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 15-17986 7 16(Kbps) Endpt(s): Alias E.164Addr src EP: BR2-RTR 4005 CallSignalAddr Port RASSignalAddr Port 10.10.110.31720 10.10.110.363495 Endpt(s): Alias E.164Addr dst EP: HQ-RTR3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.11720 10.10.110.150005 LocalCallIDAge(secs) BW 16-17986 7 128(Kbps) Endpt(s): Alias E.164Addr src EP: HQ-RTR4005 CallSignalAddr Port RASSignalAddr Port 10.10.110.11720 10.10.110.150005 Endpt(s): Alias E.164Addr dst EP: BR1-RTR 3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.21720 10.10.110.253148 When the call is originated on a SIP endpoint at BR2, the call rings at BR1, then when the SCCP endpoint at BR1 answers, it goes off-hook, gets TOH as though the BR2 side has put the call on hold. The BR2 side never acknowledges the far end answered the call and continues to ring. Examining output of “debug cch323 all” it appears as though the H245 capabilities are never completed. Now, if I change the scenario so that the codec being used between BR2 and HQ is G711ulaw, then the calls from the SIP endpoints at BR2 complete without any problem. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
I agree you need to ensure that fast start is enabled in order to facilitate SIP Early Offer - H323 Fast Start interworking. It was always my understanding that H323 fast start was enabled by default on IOS platforms. In fact, when you try to enable the command “call start fast” under voice service voip - h323, the command doesn’t show up in the output of running-config. Also, this scenario doesn’t involve UCM. It was two separate CME sites. From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Saturday, January 01, 2011 9:51 AM To: Hough, Earl Cc: Ashar Siddiqui; John Nield; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing + enable Fast Start on CUBE and CUCM trunk (enable inbound fast start or sth similar). On Sat, Jan 1, 2011 at 15:49, Miron Kobelski findko...@gmail.com wrote: Hi, SIP phones by default originate outgoing calls using SIP Early Offfer. When the next hop is H.323 trunk, you might want to try enabling Fast Start on it (voice services voip or voice class h323 - call start fast). That might solve your issue with H245 negotiation. HTH kobel On Sat, Jan 1, 2011 at 15:19, Hough, Earl earl.ho...@pcmallservices.com wrote: I never had a software MTP defined. It was always a hardware transcoder at BR2 and of course, the one collocated with the GK at HQ. I’m not having an issue calling from BR1 to BR2. The issue is when I call from a SIP endpoint at BR2 back to BR1. To recap the scenario and hopefully clear up any confusion, the SIP endpoints at BR2 are using G711ulaw as defined in the voice register pool. There is a hardware transcoder at BR2 which supports the G729r8 codec. The leg between BR2 and HQ is using G729r8. The HQ GK/CUBE reterminates the media path and sends the call on to BR1. The leg between HQ and BR1 is G711ulaw. The phone at the BR1 site is an SCCP endpoint. When the calls at BR2 originate from an SCCP endpoint, the behavior is as expected and displayed below: HQ-RTR#show gatekeeper calls Total number of active calls = 2. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 15-17986 7 16(Kbps) Endpt(s): Alias E.164Addr src EP: BR2-RTR 4005 CallSignalAddr Port RASSignalAddr Port 10.10.110.31720 10.10.110.363495 Endpt(s): Alias E.164Addr dst EP: HQ-RTR3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.11720 10.10.110.150005 LocalCallIDAge(secs) BW 16-17986 7 128(Kbps) Endpt(s): Alias E.164Addr src EP: HQ-RTR4005 CallSignalAddr Port RASSignalAddr Port 10.10.110.11720 10.10.110.150005 Endpt(s): Alias E.164Addr dst EP: BR1-RTR 3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.21720 10.10.110.253148 When the call is originated on a SIP endpoint at BR2, the call rings at BR1, then when the SCCP endpoint at BR1 answers, it goes off-hook, gets TOH as though the BR2 side has put the call on hold. The BR2 side never acknowledges the far end answered the call and continues to ring. Examining output of “debug cch323 all” it appears as though the H245 capabilities are never completed. Now, if I change the scenario so that the codec being used between BR2 and HQ is G711ulaw, then the calls from the SIP endpoints at BR2 complete without any problem. _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
I'm running out of ideas. But I think this symptoms could be related to codec mismatch. Check your dial-peers (+ remember voice class codec doesn't work with voice register pool for some IOS versions). Also you might want to have a look at DTMF settings. SIP uses RFC2833 and H.323 out of band H.245 good luck ;) kobel On Sat, Jan 1, 2011 at 16:29, Hough, Earl earl.ho...@pcmallservices.comwrote: The architecture is as follows: BR2 (CME) --(WAN)- HQ (GK/CUBE) --(WAN)- BR1 (CME) There is a transcoder defined at both the BR2 and the HQ sites and registered to the local instance of telephony-services, and I can verify that these are fully functional and being called because I am able to complete calls with SCCP endpoints using all four combinations of codecs as listed below, as well as having full use of supplementary-services. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Correct…by default when you create a new GK-controlled trunk FastStart is NOT enabled. You either need to enable it on the trunk, or change the IOS side to slow start. From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Saturday, January 01, 2011 10:18 AM To: Hough, Earl Cc: Ashar Siddiqui; John Nield; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Thanks for this, good to know. I once had similar behaviour and fixed this by enabling incoming Fast Start on CUCM. I though that it was also necessary on originating gw. On Sat, Jan 1, 2011 at 16:19, Hough, Earl earl.ho...@pcmallservices.com wrote: ecent versions of the Cisco IOS use Fast Start for calls that RSVP initiates. You might have to enable slow start for backward compatibility with older versions. When any of these occur, separate H.245 channels are opened, and the sending and receiving capabilities are negotiated the slow way.” Plus, when I look at the output of “debug cch323 all” I can tell that FastStart procedures are being initiated. _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CUCM Version on exam
I think it is 7.01 -- On Jan 1, 2011, at 11:41, George Goglidze gogli...@gmail.com wrote: Hi all, I was wondering if anyone knew the exact version in a lab. 7.0 ? 7.1 ? I know the blueprint says that any major version could be on a lab, but I was wondering if anyone knew any better. Many thanks, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Hi, SIP phones by default originate outgoing calls using SIP Early Offfer. When the next hop is H.323 trunk, you might want to try enabling Fast Start on it (voice services voip or voice class h323 - call start fast). That might solve your issue with H245 negotiation. HTH kobel On Sat, Jan 1, 2011 at 15:19, Hough, Earl earl.ho...@pcmallservices.comwrote: I never had a software MTP defined. It was always a hardware transcoder at BR2 and of course, the one collocated with the GK at HQ. I’m not having an issue calling from BR1 to BR2. The issue is when I call from a SIP endpoint at BR2 back to BR1. To recap the scenario and hopefully clear up any confusion, the SIP endpoints at BR2 are using G711ulaw as defined in the voice register pool. There is a hardware transcoder at BR2 which supports the G729r8 codec. The leg between BR2 and HQ is using G729r8. The HQ GK/CUBE reterminates the media path and sends the call on to BR1. The leg between HQ and BR1 is G711ulaw. The phone at the BR1 site is an SCCP endpoint. When the calls at BR2 originate from an SCCP endpoint, the behavior is as expected and displayed below: HQ-RTR#show gatekeeper calls Total number of active calls = 2. GATEKEEPER CALL INFO LocalCallIDAge(secs) BW 15-17986 7 16(Kbps) Endpt(s): Alias E.164Addr src EP: BR2-RTR 4005 CallSignalAddr Port RASSignalAddr Port 10.10.110.31720 10.10.110.363495 Endpt(s): Alias E.164Addr dst EP: HQ-RTR3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.11720 10.10.110.150005 LocalCallIDAge(secs) BW 16-17986 7 128(Kbps) Endpt(s): Alias E.164Addr src EP: HQ-RTR4005 CallSignalAddr Port RASSignalAddr Port 10.10.110.11720 10.10.110.150005 Endpt(s): Alias E.164Addr dst EP: BR1-RTR 3001 CallSignalAddr Port RASSignalAddr Port 10.10.110.21720 10.10.110.253148 When the call is originated on a SIP endpoint at BR2, the call rings at BR1, then when the SCCP endpoint at BR1 answers, it goes off-hook, gets TOH as though the BR2 side has put the call on hold. The BR2 side never acknowledges the far end answered the call and continues to ring. Examining output of “debug cch323 all” it appears as though the H245 capabilities are never completed. Now, if I change the scenario so that the codec being used between BR2 and HQ is G711ulaw, then the calls from the SIP endpoints at BR2 complete without any problem. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] Lab 5A
I have a problem with the calling number in lab 5A. I've configured the calling party transformations on the HQ router but for some reason it only shows the 4 digit extension when I dial 911. I have a CSS configured for the transformation and the pattern set to 1XXX. I have the CSS assigned to the gateway and the use device pool... checkbox is unchecked. If I check Use Calling Party's External Phone Number Mask on the route pattern, it displays the e164 number like it's supposed to. Any suggestions? ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] How to test AAR?
I was doing ipexpert's CCIE Voice Volume 1 Lab 6A Task 6.1 - regarding Automated Alternate Route (AAR). In the Verification section, it says Test AAR by reducing the RSVP BANDWIDTH allocated to either the HQ or BR1 WAN interface down to 39kbps, and then making a call from HQ Phone 2 to BR1 Phone 2. How exactly do I do that? I applied the command ip rsvp bandwidth 39 to HQ/BR1 serial (frame-relay) interface, but the call still goes through IP network instead of PSTN. How does CallManager know the bandwidth has been decreased? Thanks! Michael ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
The architecture is as follows: BR2 (CME) --(WAN)- HQ (GK/CUBE) --(WAN)- BR1 (CME) There is a transcoder defined at both the BR2 and the HQ sites and registered to the local instance of telephony-services, and I can verify that these are fully functional and being called because I am able to complete calls with SCCP endpoints using all four combinations of codecs as listed below, as well as having full use of supplementary-services. From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Saturday, January 01, 2011 10:11 AM To: Hough, Earl Cc: Ashar Siddiqui; John Nield; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Sorry, I don't fully understand your architecture (in your first post you indicated you want to have different codecs on both CUBE call legs). But if you want to do transcoding on CUBE you need to register transcoder on it (via telephony services). On Sat, Jan 1, 2011 at 15:36, Hough, Earl earl.ho...@pcmallservices.com wrote: After playing with codec settings on both sides of the GK/CUBE the results I am experiencing are as follows. In each of these tests I am able successfully call from a SCCP endpoint at BR2 to a SCCP endpoint at BR1. And in each of these tests, regardless of the WAN codec, the SIP endpoints are using G711ulaw as defined in “voice register pool”. BR2 (Sip Endpoint) - (G.711ulaw) -- HQ-RTR (GK/CUBE) -- (G.711ulaw) BR1*SUCCESSFUL* BR2 (Sip Endpoint) - (G729r8) -- HQ-RTR (GK/CUBE) -- (G.729r8) BR1*SUCCESSFUL* BR2 (Sip Endpoint) - (G729r8) -- HQ-RTR (GK/CUBE) -- (G.711ulaw) - BR1*NOT SUCCESSFUL* BR2 (Sip Endpoint) - (G711ulaw) --- HQ-RTR (GK/CUBE) -- (G.729r8) BR1*NOT SUCCESSFUL* So, it appears as though when both legs passing through the HQ-RTR GK/CUBE instance are the same, the calls from SIP endpoints can complete. Whenever the codecs are different, the calls cannot complete. _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ _ The information contained in this transmission is confidential. It is intended solely for the use of the individual(s) or organization(s) to whom it is addressed. Any disclosure, copying or further distribution is not permitted unless such privilege is explicitly granted in writing by PC Mall, Inc. Furthermore, PC Mall, Inc. is not responsible for the proper and complete transmission of the substance of this communication, nor for any delay in its receipt. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CCIE_Voice Digest, Vol 59, Issue 9
Review your dsp resource configuration for callmanager express.sccp should be registered with the telephony service. On Jan 1, 2011 12:50 PM, ccie_voice-requ...@onlinestudylist.com wrote: Send CCIE_Voice mailing list submissions to ccie_voice@onlinestudylist.com To subscribe or unsubscribe via the World Wide Web, visit http://onlinestudylist.com/mailman/listinfo/ccie_voice or, via email, send a message with subject or body 'help' to ccie_voice-requ...@onlinestudylist.com You can reach the person managing the list at ccie_voice-ow...@onlinestudylist.com When replying, please edit your Subject line so it is more specific than Re: Contents of CCIE_Voice digest... Today's Topics: 1. Re: CME SIP Endpoints with GK/CUBE Routing (Miron Kobelski) -- Message: 1 Date: Sat, 1 Jan 2011 16:30:10 +0100 From: Miron Kobelski findko...@gmail.com To: Hough, Earl earl.ho...@pcmallservices.com Cc: ccie_voice@onlinestudylist.com, John Nield johnni...@hotmail.com Subject: Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing Message-ID: aanlkti=twm4qmnvp_vpmh6+cx=q=33od4vpvt6wk2...@mail.gmail.com Content-Type: text/plain; charset=utf-8 I'm running out of ideas. But I think this symptoms could be related to codec mismatch. Check your dial-peers (+ remember voice class codec doesn't work with voice register pool for some IOS versions). Also you might want to have a look at DTMF settings. SIP uses RFC2833 and H.323 out of band H.245 good luck ;) kobel On Sat, Jan 1, 2011 at 16:29, Hough, Earl earl.ho...@pcmallservices.com wrote: The architecture is as follows: BR2 (CME) --(WAN)- HQ (GK/CUBE) --(WAN)- BR1 (CME) There is a transcoder defined at both the BR2 and the HQ sites and registered to the local instance of telephony-services, and I can verify that these are fully functional and being called because I am able to complete calls with SCCP endpoints using all four combinations of codecs as listed below, as well as having full use of supplementary-services. -- next part -- An HTML attachment was scrubbed... URL: /archives/ccie_voice/attachments/20110101/237a93cc/attachment-0001.html -- ___ CCIE_Voice mailing list CCIE_Voice@onlinestudylist.com http://onlinestudylist.com/mailman/listinfo/ccie_voice End of CCIE_Voice Digest, Vol 59, Issue 9 * ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] CUCM Version on exam
Hi all, I was wondering if anyone knew the exact version in a lab. 7.0 ? 7.1 ? I know the blueprint says that any major version could be on a lab, but I was wondering if anyone knew any better. Many thanks, ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco Presence Issue
Hi Shafqut, 1. is the SIP trunk created in CUCM ? 2. Do you see the SIP trunk in CUPS? 3. Check if there is any firewall between CUPC and CUPS blocking port 5060. -- probably this may be reason 4. incoming/outgoing ACL on CUPS 5. Restart CUP services. 6. Login to CUP cli and do telnet cucmip 5060 vice versa Try these first. -S _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of shafqut hamid Sent: Sunday, January 02, 2011 10:06 PM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] Cisco Presence Issue I have configured CUPS but when viewing topology page its showing red cross on server with following description * SIP Proxy (UNKNOWN) * Presence Engine (UNKNOWN) * Presence Engine Database (UNKNOWN) * CUP Database (UNKNOWN) * Sync Agent (UNKNOWN) * Inter-Cluster Sync Agent (UNKNOWN) while checking server status on CUPC its showing everything green except voicemail (Failed to Connect-No user Credential) and LDAP (Not Available-Invalid Credentials). Pictures are attached for your reference. Can someone please advise where to look to resolve these? Regards, Shamid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
If u r going from sip to sccp, definitely need transcoder on ur br1 gateway. Try it n let us know. duy ccie #27737 voice tmobile g2 On Jan 1, 2011 10:39 AM, Miron Kobelski findko...@gmail.com wrote: I'm running out of ideas. But I think this symptoms could be related to codec mismatch. Check your dial-peers (+ remember voice class codec doesn't work with voice register pool for some IOS versions). Also you might want to have a look at DTMF settings. SIP uses RFC2833 and H.323 out of band H.245 good luck ;) kobel On Sat, Jan 1, 2011 at 16:29, Hough, Earl earl.ho...@pcmallservices.com wrote: The architecture is as follows: BR2 (CME) --(WAN)- HQ (GK/CUBE) --(WAN)- BR1 (CME) There is a transcoder defined at both the BR2 and the HQ sites and registered to the local instance of telephony-services, and I can verify that these are fully functional and being called because I am able to complete calls with SCCP endpoints using all four combinations of codecs as listed below, as well as having full use of supplementary-services. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cti ports not registered with UCCX
Restarting CTI manager on CUCM resolved this issue. _ From: Shrini [mailto:linuxbos...@gmail.com] Sent: Sunday, January 02, 2011 12:40 PM To: 'Ashar Siddiqui'; 'haroon javed' Cc: 'ccie_voice@onlinestudylist.com' Subject: RE: [OSL | CCIE_Voice] cti ports not registered with UCCX I have configured UCCX atleast 20 times never had this problem. Coincidently I am have the same exact problem now. I reverted back the snapshot to clean UCCX , reconfigured step by step (formally looking to the document) ICD RPs are showing as unknown :-( Were you able to figure out what the problem is ? -Shrini _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Saturday, January 01, 2011 3:58 AM To: haroon javed Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] cti ports not registered with UCCX If you have changed anything in CTI ports at CUCM after creating them at UCCX then this issue may occur. Delete all CTI ports from CUCM, go into Route plan report and delete the numbers assigned to them as well. Do an IP telephony resync thru IPCC and I hope this may resolve the issue. Ash On 31/12/2010 05:01, haroon javed wrote: Hi, i am integrating CRS 7.0 with CUCM 6.2. when i create CTI port on UCCX, it create CTI ports on CUCM but on CUCM it is not showing that they are registered with UCCX. i also resync the jtapi on UCCX. CTI manager service is also running on Call Manager(CUCM). In UCCX it is Showing RmCm Subsystem Patial Service.(Initilization) Please help me out] thnaks -- Regards, Haroon Javed Telecom Engineer Cell: +92 (321) 8430260 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] need help for Lab-7-section 2.4 2.5 ??
Hi all, Does any body has successfully deployed task no 2.4 2.5 from Lab 7 ?? Regards ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco Presence Issue
Shrini, Thanks for your reply. 1. Yes SIP trunk is created in CUCM 2. Yes i can see the SIP trunk in CUPS and its published. 3. I tried by disabling both AntiVirus and Firewall 4. There are three incoming ACLs created automatically (AXL server ACL, all, cups ACL) 5. Restarted but no luck still show CUPS server in topology page with red cross. 6. I am unable to find telnet from CLI Still all components show UNKNOWN by moving cursor to the server with red cross on Topology page. Regards, Shafqut Hamid On Mon, Jan 3, 2011 at 1:54 PM, Shrini linuxbos...@gmail.com wrote: Hi Shafqut, 1. is the SIP trunk created in CUCM ? 2. Do you see the SIP trunk in CUPS? 3. Check if there is any firewall between CUPC and CUPS blocking port 5060. -- probably this may be reason 4. incoming/outgoing ACL on CUPS 5. Restart CUP services. 6. Login to CUP cli and do telnet cucmip 5060 vice versa Try these first. -S -- *From:* ccie_voice-boun...@onlinestudylist.com [mailto: ccie_voice-boun...@onlinestudylist.com] *On Behalf Of *shafqut hamid *Sent:* Sunday, January 02, 2011 10:06 PM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] Cisco Presence Issue I have configured CUPS but when viewing topology page its showing red cross on server with following description - SIP Proxy (UNKNOWN) - Presence Engine (UNKNOWN) - Presence Engine Database (UNKNOWN) - CUP Database (UNKNOWN) - Sync Agent (UNKNOWN) - Inter-Cluster Sync Agent (UNKNOWN) while checking server status on CUPC its showing everything green except voicemail (Failed to Connect-No user Credential) and LDAP (Not Available-Invalid Credentials). Pictures are attached for your reference. Can someone please advise where to look to resolve these? Regards, Shamid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] cti ports not registered with UCCX
I have configured UCCX atleast 20 times never had this problem. Coincidently I am have the same exact problem now. I reverted back the snapshot to clean UCCX , reconfigured step by step (formally looking to the document) ICD RPs are showing as unknown :-( Were you able to figure out what the problem is ? -Shrini _ From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Ashar Siddiqui Sent: Saturday, January 01, 2011 3:58 AM To: haroon javed Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] cti ports not registered with UCCX If you have changed anything in CTI ports at CUCM after creating them at UCCX then this issue may occur. Delete all CTI ports from CUCM, go into Route plan report and delete the numbers assigned to them as well. Do an IP telephony resync thru IPCC and I hope this may resolve the issue. Ash On 31/12/2010 05:01, haroon javed wrote: Hi, i am integrating CRS 7.0 with CUCM 6.2. when i create CTI port on UCCX, it create CTI ports on CUCM but on CUCM it is not showing that they are registered with UCCX. i also resync the jtapi on UCCX. CTI manager service is also running on Call Manager(CUCM). In UCCX it is Showing RmCm Subsystem Patial Service.(Initilization) Please help me out] thnaks -- Regards, Haroon Javed Telecom Engineer Cell: +92 (321) 8430260 ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Access
Hello ShinGei, Thanks for the info. In the end, MGCP will not make it for us [?] Regards, Roger Carpio. On Sun, Jan 2, 2011 at 9:34 AM, ShinGei Yong shingei.y...@gmail.com wrote: Hi Roger, MVA on a MGCP control gateway is possible.In fact,that is a coexisting of both MGCP and H323 on the same gateway,but you could not used a MGCP control PRI for MVA. you may refer to Netpro for own interest. https://supportforums.cisco.com/thread/2005673 Shingei. On Sun, Jan 2, 2011 at 11:16 PM, Shrini linuxbos...@gmail.com wrote: Hi ShinGei , bkvalentine, Rogers et al I remember it was successful last time when I configured it another lab when HQ was h323. Now I was confused around dial-peers hence had the question. I will give a try now with MGCP + H323 on HQ and it should work. Thanks all. Shrini *From:* ShinGei Yong [mailto:shingei.y...@gmail.com] *Sent:* Sunday, January 02, 2011 6:42 AM *To:* Shrini; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] Mobile Voice Access Hi Shrini, What about your UCM configuration? 1. is your H323 GW registered with UCM? 2. what is your dialing behavior internally?4 or 10?if is 4, then your in outbound dp should be 4 digit patten as well instead of 10. Please provide more info Shingei On Sun, Jan 2, 2011 at 4:13 PM, Shrini linuxbos...@gmail.com wrote: Hi Experts, *Wish you all a Happy and Prosperous New Year 2011* First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com 330.gif___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Cisco Presence Issue
To add one thing more that everything is green on System Troubleshooting page except the warnings for components which I haven't configured yet. Regards, Shamid Sent from my BlackBerry® smartphone using my Telenor Persona connection -Original Message- From: shafqut hamid shafqut.ha...@gmail.com Date: Mon, 3 Jan 2011 11:05:42 To: ccie_voice@onlinestudylist.com Subject: Cisco Presence Issue I have configured CUPS but when viewing topology page its showing red cross on server with following description - SIP Proxy (UNKNOWN) - Presence Engine (UNKNOWN) - Presence Engine Database (UNKNOWN) - CUP Database (UNKNOWN) - Sync Agent (UNKNOWN) - Inter-Cluster Sync Agent (UNKNOWN) while checking server status on CUPC its showing everything green except voicemail (Failed to Connect-No user Credential) and LDAP (Not Available-Invalid Credentials). Pictures are attached for your reference. Can someone please advise where to look to resolve these? Regards, Shamid ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Thanks for this, good to know. I once had similar behaviour and fixed this by enabling incoming Fast Start on CUCM. I though that it was also necessary on originating gw. On Sat, Jan 1, 2011 at 16:19, Hough, Earl earl.ho...@pcmallservices.comwrote: ecent versions of the Cisco IOS use Fast Start for calls that RSVP initiates. You might have to enable slow start for backward compatibility with older versions. When any of these occur, separate H.245 channels are opened, and the sending and receiving capabilities are negotiated the slow way.” Plus, when I look at the output of “debug cch323 all” I can tell that FastStart procedures are being initiated. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] Mobile Voice Access
Hello, Did you add this gateway as H323 in CUCM as well? Features and Services guide does not seem to mention MGCP for this feature. http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_0_1/ccmfeat/fsmobmgr.html#wp1206437 Were you specifically asked to have MVA and MGCP without H323? Regards, Roger Carpio. On Sun, Jan 2, 2011 at 2:13 AM, Shrini linuxbos...@gmail.com wrote: Hi Experts, *Wish you all a Happy and Prosperous New Year 2011* First question this year :-) HQ Site is MGCP. When I call HQ Phone 5002 , HQ PSTN is ringing -- all is good. I have configured MVA number 5999 in service parameters and Media Resources -- MVA -- 5999 / PT-INTERNAL / English on router. application service cmm http://177.1.10.10:8080/ccmivr/pages/IVRMainpage.vxml ! ! dial-peer voice 5999 pots service cmm incoming called-number 2123945999 no digit-strip Also on CUCM : But when I call 2123945999 from HQ PSTN 2123942123 I am getting fast busy tone. It is not invoking the vxml script. What am I doing wrong here ? TIA Shrini ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] CME SIP Endpoints with GK/CUBE Routing
Sorry, I missed the part where you mention these are 2 CMEs. But regarding Fast Start - I've just checked that my router never shows this command in running config. Both call start fast and no call start fast don't show up. I can't figure out any other command to verify this at the moment, but I was convinced that Fast Start is not default option. It won't hurt to try it. Let us know. RS30-CCIE(config)#voice service voip RS30-CCIE(conf-voi-serv)#h323 RS30-CCIE(conf-serv-h323)#do sh run | s h323 h323 RS30-CCIE(conf-serv-h323)#call start fast RS30-CCIE(conf-serv-h323)#do sh run | s h323 h323 RS30-CCIE(conf-serv-h323)#no call start fast RS30-CCIE(conf-serv-h323)#do sh run | s h323 h323 RS30-CCIE(config)# On Sat, Jan 1, 2011 at 15:58, Hough, Earl earl.ho...@pcmallservices.comwrote: I agree you need to ensure that fast start is enabled in order to facilitate SIP Early Offer - H323 Fast Start interworking. It was always my understanding that H323 fast start was enabled by default on IOS platforms. In fact, when you try to enable the command “call start fast” under voice service voip - h323, the command doesn’t show up in the output of running-config. Also, this scenario doesn’t involve UCM. It was two separate CME sites. ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Roig, Each call in Location based CAC : g729 - 24k g711 - 80k RSVP: g729 - 40k g711 - 96k Gatekeeper: g729 - 16k g711 - 128k (not sure 100%) For your question Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 So one call need 64k but 40 is hardcoded for rsvp. Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Just found another archive straight from Vik's mail box. In more detail http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg09933.html Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Francesc, A payload of 20 and 10 is not correct. RSVP and LLQ calculations are two different things. For RSVP, you calculations are correct. Correct Payloads (20 ms) G711 - 160 G729 - 20 For example, FRF.12, G729, with compression: IP/UDP/RTP - 2 bytes G729 - 20 bytes FRF.12 - 8 bytes 2 + 20 + 8 = 30 bytes per packet 30 bytes * 8 bits = 240 bits per packet 240 bits per packet * 50 packets per second = 12000 bits per second or 12 Kbps FRF.12, G729 without compression: IP/UDP/RTP = 40 bytes G729 - 20 bytes FRF.12 - 8 bytes 40 + 20 + 8 = 68 bytes per packet 68 * 8 = 544 bits per packets 544 bpp * 50 packets per second = 27200 bits per second or 27.2 Kbps FRF.12, G711 without compression: IP/UDP/RTP = 40 G711 = 160 FRF.12 - 8 40 + 160 + 8 = 208 bytes per packets 208 * 8 = 1664 bpp 1664 * 50 pps = 83200 bps or 83.2 Kbps Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Wednesday, January 05, 2011 7:42 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Shrini, Thank you for your answer. I don't see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 De: Shrini [mailto:linuxbos...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 18:33 Para: Roig Borrell, Francesc Xavier CC: ccie_voice@onlinestudylist.com Asunto: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Roig, Each call in Location based CAC : g729 - 24k g711 - 80k RSVP: g729 - 40k g711 - 96k Gatekeeper: g729 - 16k g711 - 128k (not sure 100%) For your question Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 So one call need 64k but 40 is hardcoded for rsvp. Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi, Take your example,my calculation for LLQ would be: (3 calls x 26.8kbps) + (1 call x 40kbps) = 120.4kbps ==121kbps I provision the first 3 calls in L2 bandwidth calculation,then i'll used L3 bandwidth calculation for the 4th call,which is the worst case. So i'll configure the PQ with above bandwidth. Shingei. On Thu, Jan 6, 2011 at 1:34 AM, Shrini linuxbos...@gmail.com wrote: Just found another archive straight from Vik's mail box. In more detail http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg09933.html Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
In rsvp for one call 40k if you set rsvp bandwidth to say 121k 3 g729 (3x40) calls will be allowed, even if you set 159 only 3 calls if you increase it to 160 all 4 calls pass It actually does not set the bandwidth, based one the number it calculates the number of calls. To test it you may try setting ip rsvp bandwidth to 79 and then next time to 80 when 79 only one call is allowed and in 80 both calls should go though. or try above b/w numbers and test number of calls. Thanks Shrini On 1/5/2011 10:26 AM, ShinGei Yong wrote: Hi, Take your example,my calculation for LLQ would be: (3 calls x 26.8kbps) + (1 call x 40kbps) = 120.4kbps ==121kbps I provision the first 3 calls in L2 bandwidth calculation,then i'll used L3 bandwidth calculation for the 4th call,which is the worst case. So i'll configure the PQ with above bandwidth. Shingei. On Thu, Jan 6, 2011 at 1:34 AM, Shrini linuxbos...@gmail.com mailto:linuxbos...@gmail.com wrote: Just found another archive straight from Vik's mail box. In more detail http://www.mail-archive.com/ccie_voice@onlinestudylist.com/msg09933.html Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visitwww.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com http://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi, RSVP reservation and actual LLQ usage are 2 different things. I think you should keep in mind, that there is no traffic in PQ before RSVP reservation completes. For RSVP calculation you only take into account L3. You have 2 possible bandwidth values: * standard (24kbps for G729/20ms) and * worst case (40 kbpbs for G729/10ms), because when the destination is ringing capabilities exchange has not yet occured and there is no media flow. That's why at this stage worst case is assumed (g729/40ms). PQ is still empty. As soon as the call is answered, capabilities are exchanged and decision about codec/payload is made - reservation can be decreased to standard 24kbps (g729/20ms). Only now the RTP flow can occur - PQ is filled up and served by LLQ (with values calculated including L2 overhead). One more thing - the task requirement is not very clear: RSVP traffic for me consists only of those several small RSVP protocol messages exchanged during RSVP negotiation. I'd not include RTP traffic in it... So I guess 5kbps should be more than enough. Anybody disagrees? HTH kobel On Wed, Jan 5, 2011 at 19:10, Roig Borrell, Francesc Xavier francesc.ro...@tecnocom.es wrote: Hi Shrini, Thank you for your answer. I don’t see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Francesc, As I noted before, the RSVP bandwidth calculation is different from the LLQ bandwidth calculation. For the scenario of 2 RSVP calls, you will need to calculate as follows: 40 + 24 = 64 (one worst case 10ms call and one normal 20 ms) So under the serial interfaces you will configure ip rsvp bandwidth 64 The question states that you need to put the RSVP traffic in the PQ. This means that the traffic will have to be marked as EF to make it into the LLQ. Under the same serial interface, enter the ip rsvp signaling ef command Now you need to calculate your BW for the LLQ. IP/UDP/RTP - 40 Payload 20 FRF.12 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload 10 FRF.12 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Wednesday, January 05, 2011 10:10 AM To: Shrini Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Shrini, Thank you for your answer. I dont see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 De: Shrini [mailto:linuxbos...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 18:33 Para: Roig Borrell, Francesc Xavier CC: ccie_voice@onlinestudylist.com Asunto: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Roig, Each call in Location based CAC : g729 - 24k g711 - 80k RSVP: g729 - 40k g711 - 96k Gatekeeper: g729 - 16k g711 - 128k (not sure 100%) For your question Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 So one call need 64k but 40 is hardcoded for rsvp. Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Jeff, Great! Then we agree with the solution for this requirement. :) Thank you very much!! De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 20:53 Para: Roig Borrell, Francesc Xavier; 'Shrini' CC: ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Francesc, As I noted before, the RSVP bandwidth calculation is different from the LLQ bandwidth calculation. For the scenario of 2 RSVP calls, you will need to calculate as follows: 40 + 24 = 64 (one worst case 10ms call and one normal 20 ms) So under the serial interfaces you will configure ip rsvp bandwidth 64 The question states that you need to put the RSVP traffic in the PQ. This means that the traffic will have to be marked as EF to make it into the LLQ. Under the same serial interface, enter the ip rsvp signaling ef command Now you need to calculate your BW for the LLQ. IP/UDP/RTP - 40 Payload - 20 FRF.12 - 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload - 10 FRF.12 - 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Wednesday, January 05, 2011 10:10 AM To: Shrini Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Shrini, Thank you for your answer. I don't see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 De: Shrini [mailto:linuxbos...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 18:33 Para: Roig Borrell, Francesc Xavier CC: ccie_voice@onlinestudylist.com Asunto: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Roig, Each call in Location based CAC : g729 - 24k g711 - 80k RSVP: g729 - 40k g711 - 96k Gatekeeper: g729 - 16k g711 - 128k (not sure 100%) For your question Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 So one call need 64k but 40 is hardcoded for rsvp. Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.comhttp://www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
I looked at the PG and they add in the calculation as I detailed in my most recent email. However, I am totally with you. The RTP/LLQ is different from the RSVP CAC and I would think that only a few extra Kbps would account for the RSVP control traffic in the PQ. Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Miron Kobelski Sent: Wednesday, January 05, 2011 10:49 AM To: Roig Borrell, Francesc Xavier Cc: ccie_voice@onlinestudylist.com; Shrini Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi, RSVP reservation and actual LLQ usage are 2 different things. I think you should keep in mind, that there is no traffic in PQ before RSVP reservation completes. For RSVP calculation you only take into account L3. You have 2 possible bandwidth values: * standard (24kbps for G729/20ms) and * worst case (40 kbpbs for G729/10ms), because when the destination is ringing capabilities exchange has not yet occured and there is no media flow. That's why at this stage worst case is assumed (g729/40ms). PQ is still empty. As soon as the call is answered, capabilities are exchanged and decision about codec/payload is made - reservation can be decreased to standard 24kbps (g729/20ms). Only now the RTP flow can occur - PQ is filled up and served by LLQ (with values calculated including L2 overhead). One more thing - the task requirement is not very clear: RSVP traffic for me consists only of those several small RSVP protocol messages exchanged during RSVP negotiation. I'd not include RTP traffic in it... So I guess 5kbps should be more than enough. Anybody disagrees? HTH kobel On Wed, Jan 5, 2011 at 19:10, Roig Borrell, Francesc Xavier francesc.ro...@tecnocom.es wrote: Hi Shrini, Thank you for your answer. I don’t see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Definitely, Im sorry I didnt understand at first J Happy studies! Jeff From: Roig Borrell, Francesc Xavier [mailto:francesc.ro...@tecnocom.es] Sent: Wednesday, January 05, 2011 12:12 PM To: givemeccievoice2...@gmail.com; 'Shrini' Cc: ccie_voice@onlinestudylist.com Subject: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Jeff, Great! Then we agree with the solution for this requirement. J Thank you very much!! De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 20:53 Para: Roig Borrell, Francesc Xavier; 'Shrini' CC: ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Francesc, As I noted before, the RSVP bandwidth calculation is different from the LLQ bandwidth calculation. For the scenario of 2 RSVP calls, you will need to calculate as follows: 40 + 24 = 64 (one worst case 10ms call and one normal 20 ms) So under the serial interfaces you will configure ip rsvp bandwidth 64 The question states that you need to put the RSVP traffic in the PQ. This means that the traffic will have to be marked as EF to make it into the LLQ. Under the same serial interface, enter the ip rsvp signaling ef command Now you need to calculate your BW for the LLQ. IP/UDP/RTP - 40 Payload 20 FRF.12 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload 10 FRF.12 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff From: ccie_voice-boun...@onlinestudylist.com [mailto:ccie_voice-boun...@onlinestudylist.com] On Behalf Of Roig Borrell, Francesc Xavier Sent: Wednesday, January 05, 2011 10:10 AM To: Shrini Cc: ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Shrini, Thank you for your answer. I dont see very clear how you take into consideration L2 header These values 40kbps (g729 10ms)/ 24kbps (g729 20ms) only consider L3+UDP/RTP+Payload. For 2 g729 calls ip rsvp bandwith= 24+40 =64 OK But which value would you use for priority queue if you have this question Between HQ-BR1 provision enough bandwidth in the priority queue for 2 calls. Any RSVP traffic should be placed into the PQ. Ensure that you provision additional amount of bandwidth in the PQ to include RSVP traffic Thanks!! Francesc Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 De: Shrini [mailto:linuxbos...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 18:33 Para: Roig Borrell, Francesc Xavier CC: ccie_voice@onlinestudylist.com Asunto: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Hi Roig, Each call in Location based CAC : g729 - 24k g711 - 80k RSVP: g729 - 40k g711 - 96k Gatekeeper: g729 - 16k g711 - 128k (not sure 100%) For your question Lets take example of one g729 call rsvp a= Worst case = (L3+PL)*8*100 = (40+10)*8*100 = 40k b= One G729 = ((L2+L3)+PL)*50*8 = ((40)+20)*8*50 = 24 L2+L3 = IP+RTP+UDP = 40 So one call need 64k but 40 is hardcoded for rsvp. Thanks Shrini On 1/5/2011 7:42 AM, Roig Borrell, Francesc Xavier wrote: Hi everyone! I am trying to understand the right way to calculate the priority value in LLQ with a RSVP configuration. I have not been able to find documentation clarifying this. So supposing HQ-BR1 4 calls g729 ip rsvp bandwitdh = 24*3 + 40 = 112 No problem with the rsvp bandwith, 3 calls with 20ms sample rate and one call with the worst case 10ms sample rate. So following this and considering FR12 . The priority queue should be calculated this way L2 7 L2 7 L3 40 L3 40 Payload 20 Payload 10 67*8*50= 26,8kbps57*8*100 = 45,6kbps LLQ priority = 28,6*3 + 45,6 = 131,4 -132 Do you agree? Is it the right way? Thanks in advance! Francesc ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
After I just agreed with you! J Below is not the RSVP calculation. That is the LLQ bandwidth calculations. After I reviewed my notes and figured out the value necessary, I referred to the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call at 20ms. I am confused as to why they do it this way. I would think that you would use the 27.2 Kbps for each call and arrive at a 55 Kbps BW in the LLQ. I agree with you that the RSVP communications will only require minimal overhead and you can just simply add a couple of Kbps to accomplish this task. Remember, the question that Francesc was referring to assumes you have RSVP configured already, and is asking you to configure the LLQ including the necessary overhead for RSVP messages. Jeff From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Wednesday, January 05, 2011 1:13 PM To: givemeccievoice2...@gmail.com Cc: Roig Borrell, Francesc Xavier; Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi guys, Yes, thinking twice it doesn’t make a lot of sense consider the call with the worst case payload (46.4) in order to adding RSVP signaling. 1 RSVP Request Dec 17 18:47:58.630: RSVP 10.10.110.2_16548-10.10.110.1_17938[0.0.0.0]: start requesting 40 kbps FF reservation for 10.10.110.2 2 RSVP update (Call established ) Dec 17 18:49:10.047: RSVP-RESV: Locally created reservation. No admission/traffic control needed Dec 17 18:49:10.047: RSVP 10.10.110.2_16510-10.10.110.1_19416[0.0.0.0]: start requesting 24 kbps FF reservation for 10.10.110.2 In fact in the first step, there isn’t RTP traffic, so in case of congestion the PQ only will have some RSVP packets. So the requirement can be achieved PQ = 27,2*2 + X, X (extra RSVP signaling traffic, as Miron we can consider 1kbps) Now, I believe we all agree!! ☺ Thanks for your help! Happy studies! Francesc De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 22:42 Para: 'Miron Kobelski' CC: Roig Borrell, Francesc Xavier; 'Shrini'; ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation After I just agreed with you! ☺ Below is not the RSVP calculation. That is the LLQ bandwidth calculations. After I reviewed my notes and figured out the value necessary, I referred to the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call at 20ms. I am confused as to why they do it this way. I would think that you would use the 27.2 Kbps for each call and arrive at a 55 Kbps BW in the LLQ. I agree with you that the RSVP communications will only require minimal overhead and you can just simply add a couple of Kbps to accomplish this task. Remember, the question that Francesc was referring to assumes you have RSVP configured already, and is asking you to configure the LLQ including the necessary overhead for RSVP messages. Jeff From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Wednesday, January 05, 2011 1:13 PM To: givemeccievoice2...@gmail.com Cc: Roig Borrell, Francesc Xavier; Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.commailto:givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Thanks for the details debugs Jeff. Just wanted to double check with you that my examples are also correct ? Thanks again Shrini On 1/5/2011 2:42 PM, Roig Borrell, Francesc Xavier wrote: Hi guys, Yes, thinking twice it doesn’t make a lot of sense consider the call with the worst case payload (46.4) in order to adding RSVP signaling. 1 RSVP Request Dec 17 18:47:58.630: RSVP 10.10.110.2_16548-10.10.110.1_17938[0.0.0.0]: start requesting 40 kbps FF reservation for 10.10.110.2 2 RSVP update (Call established ) Dec 17 18:49:10.047: RSVP-RESV: Locally created reservation. No admission/traffic control needed Dec 17 18:49:10.047: RSVP 10.10.110.2_16510-10.10.110.1_19416[0.0.0.0]: start requesting 24 kbps FF reservation for 10.10.110.2 In fact in the first step, there isn’t RTP traffic, so in case of congestion the PQ only will have some RSVP packets. So the requirement can be achieved PQ = 27,2*2 + X, X (extra RSVP signaling traffic, as Miron we can consider 1kbps) Now, I believe we all agree!! J Thanks for your help! Happy studies! Francesc *De:*givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] *Enviado el:* miércoles, 05 de enero de 2011 22:42 *Para:* 'Miron Kobelski' *CC:* Roig Borrell, Francesc Xavier; 'Shrini'; ccie_voice@onlinestudylist.com *Asunto:* RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation After I just agreed with you! J Below is not the RSVP calculation. That is the LLQ bandwidth calculations. After I reviewed my notes and figured out the value necessary, I referred to the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call at 20ms. I am confused as to why they do it this way. I would think that you would use the 27.2 Kbps for each call and arrive at a 55 Kbps BW in the LLQ. I agree with you that the RSVP communications will only require minimal overhead and you can just simply add a couple of Kbps to accomplish this task. Remember, the question that Francesc was referring to assumes you have RSVP configured already, and is asking you to configure the LLQ including the necessary overhead for RSVP messages. Jeff *From:*Miron Kobelski [mailto:findko...@gmail.com] *Sent:* Wednesday, January 05, 2011 1:13 PM *To:* givemeccievoice2...@gmail.com *Cc:* Roig Borrell, Francesc Xavier; Shrini; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.com mailto:givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Shrini, I believe you’re correct as well, but you were detailing the RSVP BW calculation not the LLQ which the question was asking. Jeff From: Shrini [mailto:linuxbos...@gmail.com] Sent: Wednesday, January 05, 2011 3:32 PM To: Roig Borrell, Francesc Xavier Cc: givemeccievoice2...@gmail.com; 'Miron Kobelski'; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Thanks for the details debugs Jeff. Just wanted to double check with you that my examples are also correct ? Thanks again Shrini On 1/5/2011 2:42 PM, Roig Borrell, Francesc Xavier wrote: Hi guys, Yes, thinking twice it doesn’t make a lot of sense consider the call with the worst case payload (46.4) in order to adding RSVP signaling. 1 RSVP Request Dec 17 18:47:58.630: RSVP 10.10.110.2_16548-10.10.110.1_17938[0.0.0.0]: start requesting 40 kbps FF reservation for 10.10.110.2 2 RSVP update (Call established ) Dec 17 18:49:10.047: RSVP-RESV: Locally created reservation. No admission/traffic control needed Dec 17 18:49:10.047: RSVP 10.10.110.2_16510-10.10.110.1_19416[0.0.0.0]: start requesting 24 kbps FF reservation for 10.10.110.2 In fact in the first step, there isn’t RTP traffic, so in case of congestion the PQ only will have some RSVP packets. So the requirement can be achieved PQ = 27,2*2 + X, X (extra RSVP signaling traffic, as Miron we can consider 1kbps) Now, I believe we all agree!! J Thanks for your help! Happy studies! Francesc De: givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.com] Enviado el: miércoles, 05 de enero de 2011 22:42 Para: 'Miron Kobelski' CC: Roig Borrell, Francesc Xavier; 'Shrini'; ccie_voice@onlinestudylist.com Asunto: RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation After I just agreed with you! J Below is not the RSVP calculation. That is the LLQ bandwidth calculations. After I reviewed my notes and figured out the value necessary, I referred to the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call at 20ms. I am confused as to why they do it this way. I would think that you would use the 27.2 Kbps for each call and arrive at a 55 Kbps BW in the LLQ. I agree with you that the RSVP communications will only require minimal overhead and you can just simply add a couple of Kbps to accomplish this task. Remember, the question that Francesc was referring to assumes you have RSVP configured already, and is asking you to configure the LLQ including the necessary overhead for RSVP messages. Jeff From: Miron Kobelski [mailto:findko...@gmail.com] Sent: Wednesday, January 05, 2011 1:13 PM To: givemeccievoice2...@gmail.com Cc: Roig Borrell, Francesc Xavier; Shrini; ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation
Hi Shrini, The example you give was the RSVP calculation.In your example you calculated all the calls (total 4 calls)in worst case scenario,which is 40kbps per calls. I'm not very agreed with your calculation, as per SRND pages 3-65 stated: To provision 4 G729 streams: (3*24) + 40 =112kbps Only Nth call will be calculated in worst case instead of 4. Also,my previous example is mean for LLQ, not for RSVP bandwidth. Thanks Shingei. On Thu, Jan 6, 2011 at 8:02 AM, givemeccievoice2...@gmail.com wrote: Hi Shrini, I believe you’re correct as well, but you were detailing the RSVP BW calculation not the LLQ which the question was asking. Jeff *From:* Shrini [mailto:linuxbos...@gmail.com] *Sent:* Wednesday, January 05, 2011 3:32 PM *To:* Roig Borrell, Francesc Xavier *Cc:* givemeccievoice2...@gmail.com; 'Miron Kobelski'; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation Thanks for the details debugs Jeff. Just wanted to double check with you that my examples are also correct ? Thanks again Shrini On 1/5/2011 2:42 PM, Roig Borrell, Francesc Xavier wrote: Hi guys, Yes, thinking twice it doesn’t make a lot of sense consider the call with the worst case payload (46.4) in order to adding RSVP signaling. 1 RSVP Request Dec 17 18:47:58.630: RSVP 10.10.110.2_16548-10.10.110.1_17938[0.0.0.0]: start requesting 40 kbps FF reservation for 10.10.110.2 2 RSVP update (Call established ) Dec 17 18:49:10.047: RSVP-RESV: Locally created reservation. No admission/traffic control needed Dec 17 18:49:10.047: RSVP 10.10.110.2_16510-10.10.110.1_19416[0.0.0.0]: start requesting 24 kbps FF reservation for 10.10.110.2 In fact in the first step, there isn’t RTP traffic, so in case of congestion the PQ only will have some RSVP packets. So the requirement can be achieved PQ = 27,2*2 + X, X (extra RSVP signaling traffic, as Miron we can consider 1kbps) Now, I believe we all agree!! J Thanks for your help! Happy studies! Francesc *De:* givemeccievoice2...@gmail.com [mailto:givemeccievoice2...@gmail.comgivemeccievoice2...@gmail.com] *Enviado el:* miércoles, 05 de enero de 2011 22:42 *Para:* 'Miron Kobelski' *CC:* Roig Borrell, Francesc Xavier; 'Shrini'; ccie_voice@onlinestudylist.com *Asunto:* RE: [OSL | CCIE_Voice] RSVP LLQ priority value calculation After I just agreed with you! J Below is not the RSVP calculation. That is the LLQ bandwidth calculations. After I reviewed my notes and figured out the value necessary, I referred to the PG. The PG calculates the PQ bandwidth by using 1 call at 10ms and 1 call at 20ms. I am confused as to why they do it this way. I would think that you would use the 27.2 Kbps for each call and arrive at a 55 Kbps BW in the LLQ. I agree with you that the RSVP communications will only require minimal overhead and you can just simply add a couple of Kbps to accomplish this task. Remember, the question that Francesc was referring to assumes you have RSVP configured already, and is asking you to configure the LLQ including the necessary overhead for RSVP messages. Jeff *From:* Miron Kobelski [mailto:findko...@gmail.com findko...@gmail.com] *Sent:* Wednesday, January 05, 2011 1:13 PM *To:* givemeccievoice2...@gmail.com *Cc:* Roig Borrell, Francesc Xavier; Shrini; ccie_voice@onlinestudylist.com *Subject:* Re: [OSL | CCIE_Voice] RSVP LLQ priority value calculation I disagree... I would never include L2 in RSVP bandwidth calculations. To see what values RSVP uses, check show ip rsvp installed in ringing and connected states. it is 40 and 24 kbps for g729. I'd say that RSVP overhead should constitute no more then 1kbps (only several small messages during RSVP negotations!) regards kobel On Wed, Jan 5, 2011 at 20:53, givemeccievoice2...@gmail.com wrote: FRF.12 – 8 40 + 20 + 8 = 68 68 bytes * 8 bits = 544 bits per packet 544 bpp * 50 pps = 272000 bps or 27.2 Kbps 2 G729 calls * 27.2 Kbps = 54.4 Kbps or roughly 55 Kbps A basic LLQ without RSVP overhead would need to have a priority 55 command. However, the question asks for you to take this extra overhead for RSVP into account. IP/UDP/RTP - 40 Payload – 10 FRF.12 – 8 40 + 10 + 8 = 58 bytes 58 * 8 = 464 bpp 464 * 100 pps = 46400 bps or 46.4 Kbps Therefore the bandwidth calculation would instead be 27.2 + 46.4 = 73.6 Kbps or 74 Kbps. Hope this helps, Jeff ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com
[OSL | CCIE_Voice] VRacks for Sale
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