Re: [OSL | CCIE_Voice] Is the CCIE voice worth anymore?
Hi Guys, Thanks for your responses I see u guys have empathized on call routing and and UC hardware for backend deployments. However Telco OTTs are coming up with directly provide these services over the cloud . Here is a disruptive analysis : http://www.slideshare.net/deanb/disruptive-analysis-web-rtc-overview-april-2013 Anyways, this might be not be so serious afterall . Just thought of brainstorming . Thanks guys for your responses again. On Tue, Aug 27, 2013 at 6:20 PM, Lakshmish NS lakshmish...@gmail.comwrote: Didn't have time to go through the video, I believe WebRTC is nothing but a Protocol, similar to SIP, H.323. Moreover, this protocol would only appeal to the Web audience, just like Skype, or Google talk. You still need to use UC hardware and their design for enterprise deployments. I mean we don't use Google talk and Skype in companies right? SIP is open source, but still Cisco uses it. As FAQ's suggest WebRTC is an open framework for the web that enables Real Time Communications in the browser. If only UC was that easy that could be implemented through browser, we didn't have to work this hard for CCIE numbers. You might want to go through this... http://www.webrtc.org/faq You've clearly misinterpreted WebRTC here.. On Tue, Aug 27, 2013 at 5:17 PM, Drake J jdrake...@gmail.com wrote: hi All, Had a troubling question hence thought of putting it out .Looking at the UC and networking trends worldwide it appears that the future of UC and collaboration is web based. Webrtc is the protocol that the world will use and individuals and organizations just need to code their requirement based on the WEBRTC. Here is the presentation that Google recently made http://www.youtube.com/watch?v=E8C8ouiXHHk Clearly many of the UC vendors are already losing out and will be losing out in year 2014. Most of the customers are already looking at reducing the cost involved in maintaining costly UC vendor networks and their networking staff . Therefore that brings me to my question is the CCIE voice worth anymore? -Drake ___ 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] LAN QoS on HWIC-4ESW
Hi Sam,My doubts is about how i'll mark packets on a remote site that are usingHWIC-4ESW to connect IPphones.I see packets hitting the policy-map onHQ's serial interface,is because I configured this.-- SW-HQ --interface FastEthernet0/14description hq ipphonemls qos trust device cisco-phonemls qos trust cos!interface FastEthernet0/10description HQ TRUNK HQ TRUNK HQmls qos trust dscp!!---Here is the config for WAN QoS.-- HQ --interface Serial0/1/0:0.1 point-to-pointbandwidth 384ip address 10.110.33.1 255.255.255.128ip ospf mtu-ignoresnmp trap link-statusframe-relay interface-dlci 201 CISCO class AutoQoS-FR-Se0/1/0:0-201 auto qos voip trustframe-relay ip rtp header-compression!!- Remote --interface Serial0/1/0:0.1 point-to-pointbandwidth 384ip address 10.110.33.2 255.255.255.128ip ospf mtu-ignoresnmp trap link-statusframe-relay interface-dlci 101 CISCO class AutoQoS-FR-Se0/1/0:0-101 auto qos voip trustframe-relay ip rtp header-compression!!---best regardsAlexOn Aug 27, 2013, at 12:33 AM, Sam Wilson wilsonc...@gmail.com wrote:You will have to either trust the dscp marking done by the switch or you will have to mark them again. Please post your router configs and may be we can get a bit more detailedHthSent from my Windows PhoneFrom: Alex Mendoza Sent: 8/26/2013 10:54 PMTo: CCIE Study Subject: [OSL | CCIE_Voice] LAN QoS on HWIC-4ESWHi All,I'm wondering about QoS on remote site.I just applied auto-qos between HQ and remote site, everything looks fine.When I run this command on both routers, in remote site I can't see packets.HQ Class-map: AutoQoS-VoIP-RTP-Trust (match-any) 11139 packets, 704260 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46)11139 packets, 704260 bytes5 minute rate 0 bps Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0REMOTE Class-map: AutoQoS-VoIP-RTP-Trust (match-any) 0 packets, 0 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46)0 packets, 0 bytes5 minute rate 0 bps Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0For remote site I don't use HWIC-4ESW (expensive) instead am using ports from 3560 (same as HQ),after I put the commands:SW(config)#interface FastEthernet0/17SW(config-if)#shutSW(config-if)# mls qos trust device cisco-phoneSW(config-if)# mls qos trust cosSW(config-if)#no shutI started to see packets on policy-map. Class-map: AutoQoS-VoIP-RTP-Trust (match-any) 186 packets, 11904 bytes 5 minute offered rate 0 bps, drop rate 0 bps Match: ip dscp ef (46)186 packets, 11904 bytes5 minute rate 0 bps Priority: 47 kbps, burst bytes 1500, b/w exceed drops: 0How I'll mark the packets on remote site to hit the policy-map in the FR link?Thanks in advancedMendoza___ 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
[OSL | CCIE_Voice] Multicast MoH.
Hi All.Is there a solution on this...GW (h323) is configured with Outbound Fast Start using IOS MTP software and is working good."Media Termination Point Required" box checked"Enable Outbound FastStart"box checked with G711u-law 64KAlso, I configure Multicast MoH for this site and is working good for calls from other IP Phones on the cluster.but PSTN calls trough this h323 GW is not, when I place the call on hold, PSTN caller hear unicast moh.To solve this issue, I need to remove MTP required form H323 CUCM config.I see this is an expected behavior, see the note from cisco doc.http://www.cisco.com/en/US/docs/voice_ip_comm/cucm/admin/7_1_2/ccmfeat/fsmoh.htmlNote The following restriction exists for multicast music on hold (MOH) when a media termination point (MTP) is invoked. When an MTP resource gets invoked in a call leg at a site that is using multicast MOH, the caller receives silence instead o music on hold. To avoid this scenario, configure unicast MOH or Tone on Hold instead of multicast MOH.Is there a trick to get multicast on a PSTN call, when "MTP required" is active on H323 GW?Any thoughts?Alex___ 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
[OSL | CCIE_Voice] RSVP getting sometime Not Enough Bandwidth
Hi I am getting Not Enough Bandwidth for some RSVP calls it is like it works mostly but some time for some call attempt i get Not enough BW? Did anyone face this issue? How to resolve it ___ 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] RSVP getting sometime Not Enough Bandwidth
Need more information. Is it as the remote site I hope. Do you have QoS configured at the remote location? From: IE Target myfrnd...@gmail.com To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Sent: Wednesday, August 28, 2013 11:38 AM Subject: [OSL | CCIE_Voice] RSVP getting sometime Not Enough Bandwidth Hi I am getting Not Enough Bandwidth for some RSVP calls it is like it works mostly but some time for some call attempt i get Not enough BW? Did anyone face this issue? How to resolve it ___ 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
