[OSL | CCIE_Voice] lab qos for cue to cucm..
anyone tell me if this is right or need changes? configure switch 1 with policies on the 3750 switch: 1: ensure cos value 5 is mapped to dscp ef /cue signal with cs3 2: in giga int 1/0/4, make sure all incoming cue signal traffic is amrked with cs3 and guarantee to 7k bandwidth. Anything in excess should be first amrked down to dsco value of 8 before being transmitted. 3. use requirements listed in the cue section to deliver teh list of protocols to be policed. On HQ Switch side: - mls qos map policed-dscp 24 to 8 mls qos map cos-dscp 0 8 16 24 32 46 48 56 mls qos class-map match-any SUB_TO_CUE match access-group name SUB_TO_CUE policy-map SUB_TO_CUE class SUB_TO_CUE set ip dscp cs3 police 8000 8000 exceed-action policed-dscp-transmit class class-default trust dscp ip access-list extended SUB_TO_CUE permit tcp host 142.100.64.12 eq 2748 host 142.1.66.253 permit tcp host 142.100.64.12 eq smtp host 142.1.66.253 permit tcp host 142.100.64.12 eq 443 host 142.1.66.253 permit tcp host 142.100.64.12 eq 8443 host 142.1.66.253 permit tcp host 142.100.64.12 eq www host 142.1.66.253 interface GigabitEthernet1/0/1 description R1 Trunk switchport trunk encapsulation dot1q switchport trunk allowed vlan 1,100,102,202 switchport mode trunk mls qos trust dscp spanning-tree portfast trunk interface GigabitEthernet1/0/3 description Publisher Port switchport mode access mls qos trust dscp spanning-tree portfast interface GigabitEthernet1/0/4 description Subscriber Port service-policy input SUB_TO_CUE switchport access vlan 100 switchport mode access spanning-tree portfast interface GigabitEthernet1/0/13 description *** IP Phones switchports switchport access vlan 202 switchport mode access switchport voice vlan 102 mls qos trust device cisco-phone mls qos trust cos spanning-tree portfast interface GigabitEthernet1/0/14 description *** IP Phones switchports switchport access vlan 202 switchport mode access switchport voice vlan 102 mls qos trust device cisco-phone mls qos trust cos spanning-tree portfast interface GigabitEthernet1/0/15 description *** IP Phones switchports switchport access vlan 202 switchport mode access switchport voice vlan 102 mls qos trust device cisco-phone mls qos trust cos spanning-tree portfast ! = On R3 router side: -- - ! class-map match-any cue_TO_sub match access-group name cue_TO_sub ! policy-map cue_TO_sub class cue_TO_sub set ip dscp cs3 police 8000 8000 exceed-action policed-dscp-transmit class class-default trust dscp ! ip access-list extended cue_TO_sub permit tcp host 142.1.66.253 eq 2748 host 142.100.64.12 permit tcp host 142.1.66.253 eq smtp host 142.100.64.12 permit tcp host 142.1.66.253 eq 443 host 142.100.64.12 permit tcp host 142.1.66.253 eq 8443 host 142.100.64.12 permit tcp host 142.1.66.253 eq www host 142.100.64.12 ! interface serial 0/1/0:0.1 point-to-point description serial port to HQ Router service-policy output cue_TO_sub ! -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] CUPC
Hello, what if you push two times the '?' key on the 7941 phone? How many rx/tx packets? On Thu, Jul 25, 2013 at 3:12 AM, Dharambir kumar varma dharambi...@gmail.com wrote: Hi Team. i have one phone CUPC over internet...and one cisco 7941 phone internal.. both registered to call manager. when i call from cupc to 7941 or viceversa,,ring out happens and when call is connected, only dead air/ No audio.. where can i check... ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Transcoding Meeting Place
Xcoder should be assigned to the resource not supporting G.729 codec On Tue, Jul 23, 2013 at 7:38 PM, William Bell b...@ucguerrilla.com wrote: My take: The MP IP gateway needs the MRGL assigned to it. -- William Bell, CCIE #38914 blog: http://ucguerrilla.com twitter: @ucguerrilla On Jul 23, 2013, at 3:11 AM, Schmitz, Daniel wrote: Hi all, a customer has the following setup. ** ** - Across the MPLS, G.729 should be used - Meeting Place is just configured for High Capacity (G.711) only*** * - CUCM has an IOS transcoder with the following configuration *dspfarm profile 3 transcode* *codec g711ulaw* *codec g711alaw* *codec g729ar8* *codec g729abr8* *codec g729r8* *codec g729br8* *maximum sessions 6* *associate application SCCP* ** ** For some reason it is not possible to call from China to the Meeting Place, as soon as I allow G.711 via the SIP trunk everything works just fine, but with G.729 the call cannot be established. Which component needs the correct MRGL for the transcoding? ** ** ** ** image002.png ** ** Do I miss anything? ** ** Regards Daniel ** ** Senior IT-Specialist Team leader Network Communication Services Managed Cloud Services ** ** -- *DIDAS Business Services GmbH* |* *Bernerstr. 38 | 60437 Frankfurt Tel.: +49 69-95022-327 | Fax: +49 69-95022-77327 | Mobil: +49 172-525 2383 Mail: daniel.schm...@didas.de | Web: www.didas.de AG Düsseldorf HRB 63231 | USt-ID-Nr.: DE811548338 | Geschäftsführer: Dirk Kiefer * * -- ** ** Der Inhalt dieser E-Mail ist vertraulich und ausschließlich für den bezeichneten Adressaten bestimmt. Wenn Sie nicht der vorgesehene Adressat dieser E-Mail oder dessen Vertreter sein sollten, so beachten Sie bitte, dass jede Form der Kenntnisnahme, Veröffentlichung, Vervielfältigung oder Weitergabe des Inhalts dieser E-Mail unzulässig ist. Wir bitten Sie, sich in diesem Fall mit dem Absender der E-Mail in Verbindung zu setzen. Wir möchten Sie außerdem darauf hinweisen, dass die Kommunikation per E-Mail über das Internet unsicher ist, da für unberechtigte Dritte grundsätzlich die Möglichkeit der Kenntnisnahme und Manipulation besteht. The information contained in this e-mail is confidential. It is intended solely for the addressee. Access to this e-mail by anyone else is unauthorized. If you are not the intended recipient, any form of disclosure, reproduction, distribution or any action taken or refrained from in reliance on it, is prohibited and may be unlawful. Please notify the sender immediately. We also like to inform you that communication via e-mail over the internet is insecure because third parties may have the possibility to access and manipulate e-mails.Hi ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] cue register issue with ccm
if configuration is fine, try to: disassociate CTIP and CTIRP from the CUE user and reassociate them On Thu, Aug 8, 2013 at 6:28 PM, Amit Sharma aryan231...@gmail.com wrote: guys.,, anyone help me as i am using ipx remote racks...and always issue to register cue with cucm... i am doing all config but it is not registering with cucm... please tell me how can fix it? -- Thanks Regard's Amit Sharma ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fwd: RSVP Max Sessions
max session software 1 will work with one call. is not correct the information about two session will be used in a pre call. On Sat, Aug 17, 2013 at 2:42 AM, IE Target myfrnd...@gmail.com wrote: That is something even i am not clear with One call = two call leg Practically if we configure maximum session software 1 the call works? some where i read that two sessions will be used pre call ?? Any comments ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com -- .:.::Marcelo Augusto::.:. MSN: harkonmose...@hotmail.com SKYPE: harkonmoseley ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
Re: [OSL | CCIE_Voice] Fwd: RSVP Max Sessions
My apologies for guessing about two sessions. I found through testing what Marcelo confirmed, sessions 1 will allow a single call. However, FOR THE LAB the number of software sessions per call is irrelevant - configure it way higher than you think you'll need (10, 100, 500), UNLESS the question tells you not to do that. If the question states don't configured DSPs that are not needed this does not apply to software MTP, as it doesn't use DSP resources. If you want to use a hardware MTP then yes you should be worried about DSPs (I would recommend using a separate transcoder and sticking with software MTP). I wanted to also confirm my prior statement about codec pass-through and understand when/why you should use it, so I did some further testing and research today. The short answer is that FOR THE LAB it does not appear to matter if you use codec pass-through, I got the same result for both with and without pass-through. (Please note, I did yet not test complicated scenarios such as a remote site phone calling to CCX which would use an MTP for RSVP then a transcoder to g711 to talk to CCX.) Either way, the CUCM region (g729) and location (bw unlimited, rsvp mandatory (with or without video desired)) must still be set properly. Personally, I don't use pass-through because it is one more variable if I need to troubleshoot and it does not help me in the lab. For the real world there are many compelling reasons to use codec pass-through (for example cisco tells you to) including fax/modem calls and sRTP, however those are not likely in the lab (I haven't seen them in any IPExpert workbooks). I expanded testing to see what effect codec pass through had on some other setups (beyond what we expect to see in a lab). For example (test 4 below), if CUCM is set to use G711 and IOS MTP has g729r8 and codec pass-through the call will setup using 96K and connect using 80K (sho ip rsvp res). Thus, codec passthrough effectively IGNORES the codec setting you have on the IOS MTP when the CUCM endpoints negotiate a codec. If the CUCM endpoints do not negotiate, then the ios mtp codec setting will kick in. Keep reading if you're bored or curious :-) --- Here's a show sccp with my config to look as sessions vs streams: HQ-RTR#sho sccp SCCP Admin State: UP Gateway Local Interface: Loopback0 IPv4 Address: 10.10.110.1 Port Number: 2000 IP Precedence: 3 User Masked Codec list: None Call Manager: 192.168.0.21, Port Number: 2000 Priority: N/A, Version: 5.0.1, Identifier: 2 Trustpoint: N/A Call Manager: 192.168.0.101, Port Number: 2000 Priority: N/A, Version: 5.0.1, Identifier: 1 Trustpoint: N/A MTP Oper State: ACTIVE - Cause Code: NONE Active Call Manager: 192.168.0.101, Port Number: 2000 TCP Link Status: CONNECTED, Profile Identifier: 2 Reported Max Streams: 1000, Reported Max OOS Streams: 0 max STREAMS 1000 (2 streams per session configured (500), which does indicate each session is one call with two streams) Supported Codec: g729r8, Maximum Packetization Period: 60 Supported Codec: rfc2833 dtmf, Maximum Packetization Period: 30 Supported Codec: rfc2833 pass-thru, Maximum Packetization Period: 30 Supported Codec: inband-dtmf to rfc2833 conversion, Maximum Packetization Period: 30 RSVP : ENABLED dspfarm profile 2 mtp codec g729r8 rsvp maximum sessions software 500 associate application SCCP ! The output above max STREAMS 1000 while I configured 500 sessions - this does indicate each session is one end-to-end call with two streams, one for each side of the mtp. See attached (in a follow up email) detailed debugs of 5 test scenarios (rsvp bandwidth was increased to allow multiple/g711 calls) *First two test scenarios are relevant to the lab (with and without pass-through RSVP works as expected)* *1: region set g711, location set unlimited bw and rsvp mandatory, ios mtp set g729* -RESULT (as expected): rsvp uses *40k during setup, 24K when call connects* -note: show sccp conn shows codec as G729, phones also show G729 in use *2: same as test 1 but adding ios mtp codec pass-through* -RESULT (as expected): rsvp uses *40k during setup, 24K when call connects* -note: show sccp conn shows codec as pass-th, phones show G729 in use *Next two tests show codec pass-through ignores the codec set in IOS MTP* *3: region set g711, location set unlimited bw and rsvp mandatory, ios mtp * *no codec set** and pass-through on* -RESULT (as expected): rsvp uses *96k during setup, 80K when call connects* -note: show sccp conn shows codec as pass-th, phones show G711 in use *4: region set g711, location set unlimited bw and rsvp mandatory, ios mtp * *set g729** and pass-through on* -RESULT: *rsvp uses 96k during setup, 80K when call connects* (ios mtp codec setting ignored because both phones negotiate g711) -note: show sccp conn shows codec as pass-th, phones show G711 in use *Last test shows how without codec pass-through the
Re: [OSL | CCIE_Voice] mva
Hi Karen, Thanks for the response. I am not sure how to use the translation. Would you mind to share in detail ? Because,I suppose, we should use the same number as PSTN line for the remote destination number(for exact match), so that when the user call from that PSTN line/number it will prompt only to enter the PIN. Thanks Ragu From: Karen Johnson karen.johnson...@yahoo.ca To: Ragulan Sathasivan s_s_ragu...@yahoo.com; ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Saturday, 17 August 2013, 0:30 Subject: Re: [OSL | CCIE_Voice] mva hi Ragu, I also confused about this if CIsco want to see use 1 or not. However not to conflict it is very easy, u just need to use Translation and change your Remote destination# Other who got 100 score on this, can pls advice? K From: Ragulan Sathasivan s_s_ragu...@yahoo.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Thursday, August 15, 2013 8:25:44 PM Subject: Re: [OSL | CCIE_Voice] mva Hi Guys, What should be the busy trigger set for the MVA IP Phone ? If I set 1 for the voicemail/Unity Connection requirement then if i call from PSNT phone line which is Remote Destination number then call is not successful. If the busy trigger changed to 2 then the call from the Remote Destination PSTN line to the MVA IP Phone is success. Regards Ragu ___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com___ For more information regarding industry leading CCIE Lab training, please visit www.ipexpert.com Are you a CCNP or CCIE and looking for a job? Check out www.PlatinumPlacement.com
