Re: [OSL | CCIE_Voice] QOS LLQ
Ryan. I got really disappointed for lack of what i call friendly approach about this part of QOS so I left it alone for a while and today i came back to give it another shot and then WOOOW Ryan you are a life saver . nice explanation pal. That was exactly where i got lost in QOS and now I am all clear. May i ask you one question about QOS marking . There are different approached to do the marking specially about the CS3 marking since new SRND recommends to remark the SIG with CS3 . I see different approaches to achieve this using class Maps 1. Match Protocols in class-map and remark with policy-map 2. have access-lists to match the packets , then match with class-map following with remarking by policy-map 3. using dial-peer commands where call is hitting the WAN directly ( for example from CME to GK to CCM ) like ip qos dscp cs3 signaling can one of these methods do the whole job of matching so we don't put time on configuring for example each and every dial-peer with CS3 . I personally think the access-lists are the one who can take care of this matter. having an access list and match any with destination port SIP or SCCP or MGCP ...and assign it to class-map and remark it with policy-map Please advice . Best of regards. Mo. On Sun, Oct 19, 2008 at 10:31 PM, Gruela, Ramil [EMAIL PROTECTED]wrote: 256 divided by total bandwidth maybe? I don't know. Can't assign priority to multiple classes. *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mo *Sent:* Thursday, October 16, 2008 10:59 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] QOS LLQ Hi everybody . hope you all having a good day . got a question about QOS LLQ . as far as i understand to reserve amount of bandwidth for media and control where media=256kbps and SIG=8 kbps Class RTP priority 256--- 256 Kbps of which bandwidth, Interface or CIR or MINCIR Class SIG Bandwidth 8 --- same as above question also what if we have to allocate percentage of bandwidth in the above mentioned scenario what would be the correct percentage to use with eg. allocate VOICE=256kbps and SIG=8kbps . Class RTP priority percent--- what to calculate for this Class SIG priority Precent---same as above question basically i,m lost in calculation of these parameters.
Re: [OSL | CCIE_Voice] what is this suppose to mean? different Codecs show up as G711 and G729 on the call parties for a call,
Hi , So, What I'm understanding is u say calls is routed to BR2 in G729 and BR2 changes codec to G711 to send it to IPphones. some point that I cannot understand if this is the case. -How can BR2 (gateway) do transcoding? since I didn't set up any transcoding on that so it should not use dspfarm for transcoding. -PSTN is not an IPIPGW so calls not ended on pstn and then pstn make them again. so I should see G729 at least on pstn show call active voice, but what I see is G711. one of call legs is in HQ with G729 and another one is in PSTN with G711 !!! how is this possible? - Base on your statement ,I should see G729 coming to BR2 and there should be a way to see this behavior on Br2 to verify it - if assume this is the case and Br2 convert it to G711 then how can Br2 receive calls in G729 and I see G711 on Pstn?? Jeremy On Thu, Oct 23, 2008 at 3:26 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: The origination codec has nothing to do with the codec in BR2, for example, in real live the call could be generated in GSM but it is always going to be delivered in G711 at BR2 to internal Phones. R2 and ISDN use 64 Kbps channels to send the call in a digital format, and the receiving router takes that call and send it in G711 to ip phones and G729 to any other voip dial-peer. And, yes, the gateways need DSPs to perform kind of transcoding to adapt any internal codec to the one used by the PSTN media, no matter if it is FXS, FXO, EM, R2, ISDN etc. Rgds//r.a. On Wed, Oct 22, 2008 at 12:01 PM, jeremy co [EMAIL PROTECTED]wrote: so my pstn transcode my calls? software transcoding? How can I prevent that? in actual ccie lab ,they should do something to transfer original call codec. so if I configure codec transparent it would be OK? Jeremy On Thu, Oct 23, 2008 at 2:51 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Ok, so in this case, the call from HQ to GK is a G729 call as defined by your region. Then the call is routed to PSTN by the GK. The BR2 receive tha call as G711 as default a codec and send it to any internal phone. All the call coming from PSTN are handled by default as G711 calls. //r.a. On Wed, Oct 22, 2008 at 11:39 AM, jeremy co [EMAIL PROTECTED]wrote: my PSTN is 2600 router with 2xT1 and 2xE1, link between Br2 and pstn is E1---R2 signaling and between HQ and pstn is T1,PRI Br2 not registered to GK, just HQ and CCM registered to GK Tanx Jeremy On Thu, Oct 23, 2008 at 2:32 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Jeremy, the call in this segment: ...GK/PSTN-BR2 (CME)... Is actually a PSTN call? i mean are you using any ISDN or R2 line? Or Is it an Gatekeeper handled call? (BR2 registered to GK) //r.a. On Wed, Oct 22, 2008 at 4:20 AM, jeremy co [EMAIL PROTECTED]wrote: Hi, suppose following scenario: IPPHONE1 CCM--HQGK/PSTN-BR2 (CME)IPPHONE 2 I force CCM to use G729, by creating region using G729. call coming from ip phone1 to GK via trunk from CCM and GK routes it to HQ. HQ transfer it to PSTN and then call goes to BR2 and IPphone 2. On IPPHONE1 -G729 on HQ ---using show call active voiceG729 on PSTN router --G711 on IPPHONE2 G711 Why I see two different codec on PSTN router and IP phone1? How come codec changes from G729 to G711? How can I prevent it and see G729 on IPphone2 as well? Jeremy
[OSL | CCIE_Voice] SRST 4-digits preservation
I know I asked this bfore, but still didn't get warm and fuzzy feeling about this. The question: We need to preserve 4-digits dialing between HQ and Br1 during SRST. On the SRST side it's easy - create POTS dial-peer with destination-pattern 2... with prefix. However it's still not clear how to preserver 4-digits dialing from HQ to SRST site. So far I tried different things: 1. Create translation pattern for numbers, place it in a partition, and play with an order of partitions in CSS. If the partition is above internal partition, then all calls go to full E.164 number with or without SRST. 2. Same thing with CTI RP. 3. Creating call forward on no coverage does not do anything. When phone unregistered on the callmanager, it gets either reorder tone (if no VM configured), or goes directly to VM. 4. So far the only option works - configure phone with forward no answer to full E.164 address. However this will invalidate the requirement of having phone forwarded to VM when nobody answers. So, any thoughts on this? Somehow all workbooks I saw ignore this part - they all describe how to configure SRST router, but nothing on how to configure CallManager... Good luck, Mike
Re: [OSL | CCIE_Voice] what is this suppose to mean? different Codecs show up as G711 and G729 on the call parties for a call,
The call is routed as such: UCM HQ Phone(VoIP G729)--HQ/GK (VoIP G729)---PSTN-(NO LONGER VOIP - NOW PCM THROUGH E1-R2 CAS) ---BR2(VoIP G711)IPPhone. Bottom line is that once the PSTN has it - it enters BR2 *not* as G711 - in fact *not* as VOIP at all - it is POTS. Then the DSPs in BR2 convert POTS PCM to G711 and send it along to the BR2 IP Phone. I hope this helps a little. -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 22, 2008, at 12:49 PM, jeremy co wrote: Hi , So, What I'm understanding is u say calls is routed to BR2 in G729 and BR2 changes codec to G711 to send it to IPphones. some point that I cannot understand if this is the case. -How can BR2 (gateway) do transcoding? since I didn't set up any transcoding on that so it should not use dspfarm for transcoding. -PSTN is not an IPIPGW so calls not ended on pstn and then pstn make them again. so I should see G729 at least on pstn show call active voice, but what I see is G711. one of call legs is in HQ with G729 and another one is in PSTN with G711 !!! how is this possible? - Base on your statement ,I should see G729 coming to BR2 and there should be a way to see this behavior on Br2 to verify it - if assume this is the case and Br2 convert it to G711 then how can Br2 receive calls in G729 and I see G711 on Pstn?? Jeremy On Thu, Oct 23, 2008 at 3:26 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: The origination codec has nothing to do with the codec in BR2, for example, in real live the call could be generated in GSM but it is always going to be delivered in G711 at BR2 to internal Phones. R2 and ISDN use 64 Kbps channels to send the call in a digital format, and the receiving router takes that call and send it in G711 to ip phones and G729 to any other voip dial-peer. And, yes, the gateways need DSPs to perform kind of transcoding to adapt any internal codec to the one used by the PSTN media, no matter if it is FXS, FXO, EM, R2, ISDN etc. Rgds//r.a. On Wed, Oct 22, 2008 at 12:01 PM, jeremy co [EMAIL PROTECTED] wrote: so my pstn transcode my calls? software transcoding? How can I prevent that? in actual ccie lab ,they should do something to transfer original call codec. so if I configure codec transparent it would be OK? Jeremy On Thu, Oct 23, 2008 at 2:51 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Ok, so in this case, the call from HQ to GK is a G729 call as defined by your region. Then the call is routed to PSTN by the GK. The BR2 receive tha call as G711 as default a codec and send it to any internal phone. All the call coming from PSTN are handled by default as G711 calls. //r.a. On Wed, Oct 22, 2008 at 11:39 AM, jeremy co [EMAIL PROTECTED] wrote: my PSTN is 2600 router with 2xT1 and 2xE1, link between Br2 and pstn is E1---R2 signaling and between HQ and pstn is T1,PRI Br2 not registered to GK, just HQ and CCM registered to GK Tanx Jeremy On Thu, Oct 23, 2008 at 2:32 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Jeremy, the call in this segment: ...GK/PSTN-BR2 (CME)... Is actually a PSTN call? i mean are you using any ISDN or R2 line? Or Is it an Gatekeeper handled call? (BR2 registered to GK) //r.a. On Wed, Oct 22, 2008 at 4:20 AM, jeremy co [EMAIL PROTECTED] wrote: Hi, suppose following scenario: IPPHONE1 CCM--HQGK/PSTN-BR2 (CME) IPPHONE 2 I force CCM to use G729, by creating region using G729. call coming from ip phone1 to GK via trunk from CCM and GK routes it to HQ. HQ transfer it to PSTN and then call goes to BR2 and IPphone 2. On IPPHONE1 -G729 on HQ ---using show call active voiceG729 on PSTN router --G711 on IPPHONE2 G711 Why I see two different codec on PSTN router and IP phone1? How come codec changes from G729 to G711? How can I prevent it and see G729 on IPphone2 as well? Jeremy
Re: [OSL | CCIE_Voice] QOS LLQ
Percent in a Policy map is a derivative of Mincir but only IF IF there is a traffic shaper present on the interface. If there is no traffic shaper present - then Percent is a derivative of the bandwidth command. HTH, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 24, 2008, at 9:25 AM, Mo wrote: Ryan. I got really disappointed for lack of what i call friendly approach about this part of QOS so I left it alone for a while and today i came back to give it another shot and then WOOOW Ryan you are a life saver . nice explanation pal. That was exactly where i got lost in QOS and now I am all clear. May i ask you one question about QOS marking . There are different approached to do the marking specially about the CS3 marking since new SRND recommends to remark the SIG with CS3 . I see different approaches to achieve this using class Maps 1. Match Protocols in class-map and remark with policy-map 2. have access-lists to match the packets , then match with class- map following with remarking by policy-map 3. using dial-peer commands where call is hitting the WAN directly ( for example from CME to GK to CCM ) like ip qos dscp cs3 signaling can one of these methods do the whole job of matching so we don't put time on configuring for example each and every dial-peer with CS3 . I personally think the access-lists are the one who can take care of this matter. having an access list and match any with destination port SIP or SCCP or MGCP ...and assign it to class-map and remark it with policy-map Please advice . Best of regards. Mo. On Sun, Oct 19, 2008 at 10:31 PM, Gruela, Ramil [EMAIL PROTECTED] wrote: 256 divided by total bandwidth maybe? I don't know. Can't assign priority to multiple classes. From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] ] On Behalf Of Mo Sent: Thursday, October 16, 2008 10:59 AM To: ccie_voice@onlinestudylist.com Subject: [OSL | CCIE_Voice] QOS LLQ Hi everybody . hope you all having a good day . got a question about QOS LLQ . as far as i understand to reserve amount of bandwidth for media and control where media=256kbps and SIG=8 kbps Class RTP priority 256--- 256 Kbps of which bandwidth, Interface or CIR or MINCIR Class SIG Bandwidth 8 --- same as above question also what if we have to allocate percentage of bandwidth in the above mentioned scenario what would be the correct percentage to use with eg. allocate VOICE=256kbps and SIG=8kbps . Class RTP priority percent--- what to calculate for this Class SIG priority Precent---same as above question basically i,m lost in calculation of these parameters.
Re: [OSL | CCIE_Voice] QOS LLQ
Mark. it is either me as a dummy or it,s the subject getting way too complicated . but in either way i appreciate if you can explain it with an example . Here i got one : There a link between HQ and BR1 , bandwidth is 512 and we want to reserve 10% for voice and 5% for SIG . I have to achieve this using bandwidth command . First I have the bandwidth 512 under serial int connected to BR1. and yes i don't have any traffic shaper percent command. then I go to policy-map VOIP-RTP bandwidth X . where X = 10% of 512 ( comes from bandwidth command set under serial interface) in this case === (512 * 10% = 51Kbps rounded ). class VOIP_SIG bandwidth X . where X = 5% of 512 ( comes from bandwidth command set under serial interface) in this case === (512 * 5% = 25Kbps rounded ). class class-default fair-queue raise one question on my mind and that,s the Cisco recommendation to always reserver 5% for over head. Again thank you guys and appreciate your comments on this one . Regards. Mo. On Fri, Oct 24, 2008 at 7:28 PM, Mark Snow [EMAIL PROTECTED] wrote: Percent in a Policy map is a derivative of Mincir but only IF IF there is a traffic shaper present on the interface. If there is no traffic shaper present - then Percent is a derivative of the bandwidth command. HTH, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 24, 2008, at 9:25 AM, Mo wrote: Ryan. I got really disappointed for lack of what i call friendly approach about this part of QOS so I left it alone for a while and today i came back to give it another shot and then WOOOW Ryan you are a life saver . nice explanation pal. That was exactly where i got lost in QOS and now I am all clear. May i ask you one question about QOS marking . There are different approached to do the marking specially about the CS3 marking since new SRND recommends to remark the SIG with CS3 . I see different approaches to achieve this using class Maps 1. Match Protocols in class-map and remark with policy-map 2. have access-lists to match the packets , then match with class-map following with remarking by policy-map 3. using dial-peer commands where call is hitting the WAN directly ( for example from CME to GK to CCM ) like ip qos dscp cs3 signaling can one of these methods do the whole job of matching so we don't put time on configuring for example each and every dial-peer with CS3 . I personally think the access-lists are the one who can take care of this matter. having an access list and match any with destination port SIP or SCCP or MGCP ...and assign it to class-map and remark it with policy-map Please advice . Best of regards. Mo. On Sun, Oct 19, 2008 at 10:31 PM, Gruela, Ramil [EMAIL PROTECTED]wrote: 256 divided by total bandwidth maybe? I don't know. Can't assign priority to multiple classes. *From:* [EMAIL PROTECTED] [mailto: [EMAIL PROTECTED] *On Behalf Of *Mo *Sent:* Thursday, October 16, 2008 10:59 AM *To:* ccie_voice@onlinestudylist.com *Subject:* [OSL | CCIE_Voice] QOS LLQ Hi everybody . hope you all having a good day . got a question about QOS LLQ . as far as i understand to reserve amount of bandwidth for media and control where media=256kbps and SIG=8 kbps Class RTP priority 256--- 256 Kbps of which bandwidth, Interface or CIR or MINCIR Class SIG Bandwidth 8 --- same as above question also what if we have to allocate percentage of bandwidth in the above mentioned scenario what would be the correct percentage to use with eg. allocate VOICE=256kbps and SIG=8kbps . Class RTP priority percent--- what to calculate for this Class SIG priority Precent---same as above question basically i,m lost in calculation of these parameters.
Re: [OSL | CCIE_Voice] QOS LLQ
If there were no traffic shaper present in the task requirements (as you state here) - then yes bandwidth command must be used on the Serial interface to 512 and the bandwidth command in the policy map would be set to 51.2 or 51 or 52 since 51.2 wouldn't be accepted by IOS (ask the proctor how to round up or down). Don't worry about what Cisco recommends in the lab UNLESS you are given a specific task that states to use Cisco best practices for something. Also keep in mind - that what I stated above about bandwidth is correct and fine for a PHYSICAL interface (e.g. Serial0/1/0) but if you are on a Sub-Interface (e.g. Seria0/1/0.2) then you MUST MUST use a traffic shaper - else IOS won't let you put LLQ directly on a sub- interface without a traffic shaper. And no - you aren't a dummy - but yes - QoS can be quite complicated. But you will master it young Jedi :) -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On- Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 24, 2008, at 12:58 PM, Mo wrote: Mark. it is either me as a dummy or it,s the subject getting way too complicated . but in either way i appreciate if you can explain it with an example . Here i got one : There a link between HQ and BR1 , bandwidth is 512 and we want to reserve 10% for voice and 5% for SIG . I have to achieve this using bandwidth command . First I have the bandwidth 512 under serial int connected to BR1. and yes i don't have any traffic shaper percent command. then I go to policy-map VOIP-RTP bandwidth X . where X = 10% of 512 ( comes from bandwidth command set under serial interface) in this case === (512 * 10% = 51Kbps rounded ). class VOIP_SIG bandwidth X . where X = 5% of 512 ( comes from bandwidth command set under serial interface) in this case === (512 * 5% = 25Kbps rounded ). class class-default fair-queue raise one question on my mind and that,s the Cisco recommendation to always reserver 5% for over head. Again thank you guys and appreciate your comments on this one . Regards. Mo. On Fri, Oct 24, 2008 at 7:28 PM, Mark Snow [EMAIL PROTECTED] wrote: Percent in a Policy map is a derivative of Mincir but only IF IF there is a traffic shaper present on the interface. If there is no traffic shaper present - then Percent is a derivative of the bandwidth command. HTH, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video- On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 24, 2008, at 9:25 AM, Mo wrote: Ryan. I got really disappointed for lack of what i call friendly approach about this part of QOS so I left it alone for a while and today i came back to give it another shot and then WOOOW Ryan you are a life saver . nice explanation pal. That was exactly where i got lost in QOS and now I am all clear. May i ask you one question about QOS marking . There are different approached to do the marking specially about the CS3 marking since new SRND recommends to remark the SIG with CS3 . I see different approaches to achieve this using class Maps 1. Match Protocols in class-map and remark with policy-map 2. have access-lists to match the packets , then match with class- map following with remarking by policy-map 3. using dial-peer commands where call is hitting the WAN directly ( for example from CME to GK to CCM ) like ip qos dscp cs3 signaling can one of these methods do the whole job of matching so we don't put time on configuring for example each and every dial-peer with CS3 . I personally think the access-lists are the one who can take care of this matter. having an access list and match any with destination port SIP or SCCP or MGCP ...and assign it to class-map and remark it with
Re: [OSL | CCIE_Voice] what is this suppose to mean? different Codecs show up as G711 and G729 on the call parties for a call,
Hi Mark, Please correct me if I'm wrong POTS to POTS calls use G711 PCM codec.on all channels. Also we can verify it on pstn router by sh active call voice Telephony call-legs: 2 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 11E3 : 21 10691280ms.1 +16640 pid:200 Answer 3001 active dur 00:00:37 tx:1007/167981 rx:1373/216734 Tele 1/0:23 (21) [1/0.3] tx:47105/27165/0ms g711ulaw noise:-57 acom:6 i/0:-41/-46 dBm 11E3 : 22 10691730ms.1 +16190 pid: Originate active dur 00:00:37 tx:1373/227718 rx:1007/159925 Tele 1/2:0 (22) [1/2.2] tx:47115/20015/0ms g711ulaw noise:-44 acom:6 i/0:-44/-41 dBm Telephony call-legs: 2 SIP call-legs: 0 H323 call-legs: 0 Call agent controlled call-legs: 0 SCCP call-legs: 0 Multicast call-legs: 0 Total call-legs: 2 Jeremy On Sat, Oct 25, 2008 at 4:08 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Hi Mark, pls correct me if i'm wrong... I for sure agree the path b2n PSTN router and BR2 is not VoIP, no doubt about it... but it is G711 aka g711 (ulaw or alaw). The function of the DSPs is to take the information from the digital channel decode it and set it ready for the processor to packetize it (if VoIP), and lets suppose it is dial-peer G711 so it would be G711 again, but with headers IP and maybe another voltage levels, depending the electrical media ex: ethernet, wan, fo, etc No doubt it is a Pots-to-Pots call, the output of a show voice call status for a incoming call from pstn to internal phone is this: SiteC#sh voice call status CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers 0x511D8 0x8595A4A4 1/0:23.1 1/13:1 4001 g711ulaw 1/20001 0x611D8 0x85CCA2B4 50/0/1.0*4001 g711ulaw 20001/1 1 active call found //r.a. On Fri, Oct 24, 2008 at 11:56 AM, Mark Snow [EMAIL PROTECTED] wrote: The call is routed as such: UCM HQ Phone(VoIP G729)--HQ/GK(VoIP G729)---PSTN-(NO LONGER VOIP - NOW PCM THROUGH E1-R2 CAS) ---BR2(VoIP G711)IPPhone. Bottom line is that once the PSTN has it - it enters BR2 *not* as G711 - in fact *not* as VOIP at all - it is POTS. Then the DSPs in BR2 convert POTS PCM to G711 and send it along to the BR2 IP Phone. I hope this helps a little. -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 22, 2008, at 12:49 PM, jeremy co wrote: Hi , So, What I'm understanding is u say calls is routed to BR2 in G729 and BR2 changes codec to G711 to send it to IPphones. some point that I cannot understand if this is the case. -How can BR2 (gateway) do transcoding? since I didn't set up any transcoding on that so it should not use dspfarm for transcoding. -PSTN is not an IPIPGW so calls not ended on pstn and then pstn make them again. so I should see G729 at least on pstn show call active voice, but what I see is G711. one of call legs is in HQ with G729 and another one is in PSTN with G711 !!! how is this possible? - Base on your statement ,I should see G729 coming to BR2 and there should be a way to see this behavior on Br2 to verify it - if assume this is the case and Br2 convert it to G711 then how can Br2 receive calls in G729 and I see G711 on Pstn?? Jeremy On Thu, Oct 23, 2008 at 3:26 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: The origination codec has nothing to do with the codec in BR2, for example, in real live the call could be generated in GSM but it is always going to be delivered in G711 at BR2 to internal Phones. R2 and ISDN use 64 Kbps channels to send the call in a digital format, and the receiving router takes that call and send it in G711 to ip phones and G729 to any other voip dial-peer. And, yes, the gateways need DSPs to perform kind of transcoding to adapt any internal codec to the one used by the PSTN media, no matter if it is FXS, FXO, EM, R2, ISDN etc. Rgds//r.a. On Wed, Oct 22, 2008 at 12:01 PM, jeremy co [EMAIL PROTECTED]wrote: so my pstn transcode my calls? software transcoding? How can I prevent that? in actual ccie lab ,they should do something to transfer original call codec. so if I configure codec transparent it would be OK? Jeremy On Thu, Oct 23, 2008 at 2:51 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Ok, so in this case, the call from HQ to GK is a G729 call as defined by your
Re: [OSL | CCIE_Voice] what is this suppose to mean? different Codecs show up as G711 and G729 on the call parties for a call,
Hi Mark, pls correct me if i'm wrong... I for sure agree the path b2n PSTN router and BR2 is not VoIP, no doubt about it... but it is G711 aka g711 (ulaw or alaw). The function of the DSPs is to take the information from the digital channel decode it and set it ready for the processor to packetize it (if VoIP), and lets suppose it is dial-peer G711 so it would be G711 again, but with headers IP and maybe another voltage levels, depending the electrical media ex: ethernet, wan, fo, etc No doubt it is a Pots-to-Pots call, the output of a show voice call status for a incoming call from pstn to internal phone is this: SiteC#sh voice call status CallID CID ccVdb Port DSP/Ch Called # Codec Dial-peers 0x511D8 0x8595A4A4 1/0:23.1 1/13:1 4001 g711ulaw 1/20001 0x611D8 0x85CCA2B4 50/0/1.0*4001 g711ulaw 20001/1 1 active call found //r.a. On Fri, Oct 24, 2008 at 11:56 AM, Mark Snow [EMAIL PROTECTED] wrote: The call is routed as such: UCM HQ Phone(VoIP G729)--HQ/GK(VoIP G729)---PSTN-(NO LONGER VOIP - NOW PCM THROUGH E1-R2 CAS) ---BR2(VoIP G711)IPPhone. Bottom line is that once the PSTN has it - it enters BR2 *not* as G711 - in fact *not* as VOIP at all - it is POTS. Then the DSPs in BR2 convert POTS PCM to G711 and send it along to the BR2 IP Phone. I hope this helps a little. -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities http://www.ipexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 22, 2008, at 12:49 PM, jeremy co wrote: Hi , So, What I'm understanding is u say calls is routed to BR2 in G729 and BR2 changes codec to G711 to send it to IPphones. some point that I cannot understand if this is the case. -How can BR2 (gateway) do transcoding? since I didn't set up any transcoding on that so it should not use dspfarm for transcoding. -PSTN is not an IPIPGW so calls not ended on pstn and then pstn make them again. so I should see G729 at least on pstn show call active voice, but what I see is G711. one of call legs is in HQ with G729 and another one is in PSTN with G711 !!! how is this possible? - Base on your statement ,I should see G729 coming to BR2 and there should be a way to see this behavior on Br2 to verify it - if assume this is the case and Br2 convert it to G711 then how can Br2 receive calls in G729 and I see G711 on Pstn?? Jeremy On Thu, Oct 23, 2008 at 3:26 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: The origination codec has nothing to do with the codec in BR2, for example, in real live the call could be generated in GSM but it is always going to be delivered in G711 at BR2 to internal Phones. R2 and ISDN use 64 Kbps channels to send the call in a digital format, and the receiving router takes that call and send it in G711 to ip phones and G729 to any other voip dial-peer. And, yes, the gateways need DSPs to perform kind of transcoding to adapt any internal codec to the one used by the PSTN media, no matter if it is FXS, FXO, EM, R2, ISDN etc. Rgds//r.a. On Wed, Oct 22, 2008 at 12:01 PM, jeremy co [EMAIL PROTECTED]wrote: so my pstn transcode my calls? software transcoding? How can I prevent that? in actual ccie lab ,they should do something to transfer original call codec. so if I configure codec transparent it would be OK? Jeremy On Thu, Oct 23, 2008 at 2:51 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Ok, so in this case, the call from HQ to GK is a G729 call as defined by your region. Then the call is routed to PSTN by the GK. The BR2 receive tha call as G711 as default a codec and send it to any internal phone. All the call coming from PSTN are handled by default as G711 calls. //r.a. On Wed, Oct 22, 2008 at 11:39 AM, jeremy co [EMAIL PROTECTED]wrote: my PSTN is 2600 router with 2xT1 and 2xE1, link between Br2 and pstn is E1---R2 signaling and between HQ and pstn is T1,PRI Br2 not registered to GK, just HQ and CCM registered to GK Tanx Jeremy On Thu, Oct 23, 2008 at 2:32 AM, Ricardo Arevalo [EMAIL PROTECTED] wrote: Jeremy, the call in this segment: ...GK/PSTN-BR2 (CME)... Is actually a PSTN call? i mean are you using any ISDN or R2 line? Or Is it an Gatekeeper handled call? (BR2 registered to GK) //r.a. On Wed, Oct 22, 2008 at 4:20 AM, jeremy co [EMAIL PROTECTED] wrote: Hi, suppose following scenario: IPPHONE1
[OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config
Hi all, I have configured the phone ports on BR1 as follows, they get an IP from the DHCP scope on CUCM server which I can see also with CDP on the router. However they are not contactable via IP, if I clear the CDP table the addresses disapear. I can ping the default gateway, it is correct in DHCP as is the net mask. I sometimes get a response via ping but probably a couple out of the blue now and again. The VLAN interfaces's are set correctly because I can see them ok. interface FastEthernet1/0 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 interface FastEthernet1/8 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 I configured switchport trunk encapsulation dot1q but it does not show in the config I also tried just as an access port with access vlan and voice vlan, also without success. This seems such an easy issue as I have never had such issues before when configuring similar ? I know my brain is swimming at the moment, first lab attempt on 6th Nov, but I do not think I have missed anything? I had the same problem on the same pod yesterday.
Re: [OSL | CCIE_Voice] QOS LLQ
Thank you Mike . You not only answered my question but also answered the other questions that i,m sure i was going to face later on . specially about the Frame relay shaping command on physical interface cheers .:) //Mo On Fri, Oct 24, 2008 at 8:39 PM, Mark Snow [EMAIL PROTECTED] wrote: If there were no traffic shaper present in the task requirements (as you state here) - then yes bandwidth command must be used on the Serial interface to 512 and the bandwidth command in the policy map would be set to 51.2 or 51 or 52 since 51.2 wouldn't be accepted by IOS (ask the proctor how to round up or down). Don't worry about what Cisco recommends in the lab UNLESS you are given a specific task that states to use Cisco best practices for something. Also keep in mind - that what I stated above about bandwidth is correct and fine for a PHYSICAL interface (e.g. Serial0/1/0) but if you are on a Sub-Interface (e.g. Seria0/1/0.2) then you MUST MUST use a traffic shaper - else IOS won't let you put LLQ directly on a sub-interface without a traffic shaper. And no - you aren't a dummy - but yes - QoS can be quite complicated. But you will master it young Jedi :) -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 24, 2008, at 12:58 PM, Mo wrote: Mark. it is either me as a dummy or it,s the subject getting way too complicated . but in either way i appreciate if you can explain it with an example . Here i got one : There a link between HQ and BR1 , bandwidth is 512 and we want to reserve 10% for voice and 5% for SIG . I have to achieve this using bandwidth command . First I have the bandwidth 512 under serial int connected to BR1. and yes i don't have any traffic shaper percent command. then I go to policy-map VOIP-RTP bandwidth X . where X = 10% of 512 ( comes from bandwidth command set under serial interface) in this case === (512 * 10% = 51Kbps rounded ). class VOIP_SIG bandwidth X . where X = 5% of 512 ( comes from bandwidth command set under serial interface) in this case === (512 * 5% = 25Kbps rounded ). class class-default fair-queue raise one question on my mind and that,s the Cisco recommendation to always reserver 5% for over head. Again thank you guys and appreciate your comments on this one . Regards. Mo. On Fri, Oct 24, 2008 at 7:28 PM, Mark Snow [EMAIL PROTECTED] wrote: Percent in a Policy map is a derivative of Mincir but only IF IF there is a traffic shaper present on the interface. If there is no traffic shaper present - then Percent is a derivative of the bandwidth command. HTH, -- Mark Snow CCIE #14073 (Voice, Security) Senior Technical Instructor - IPexpert, Inc. Telephone: +1.810.326.1444 Fax: +1.309.413.4097 Mailto: [EMAIL PROTECTED] -- Join our free online support and peer group communities: http://www.IPexpert.com/communities -- IPexpert - The Global Leader in Self-Study, Classroom-Based, Video-On-Demand and Audio Certification Training Tools for the Cisco CCIE RS Lab, CCIE Security Lab, CCIE Service Provider Lab , CCIE Voice Lab and CCIE Storage Lab Certifications. -- On Oct 24, 2008, at 9:25 AM, Mo wrote: Ryan. I got really disappointed for lack of what i call friendly approach about this part of QOS so I left it alone for a while and today i came back to give it another shot and then WOOOW Ryan you are a life saver . nice explanation pal. That was exactly where i got lost in QOS and now I am all clear. May i ask you one question about QOS marking . There are different approached to do the marking specially about the CS3 marking since new SRND recommends to remark the SIG with CS3 . I see different approaches to achieve this using class Maps 1. Match Protocols in class-map and remark with policy-map 2. have access-lists to match the packets , then match with class-map following with remarking by policy-map 3. using dial-peer commands where call is hitting the WAN directly ( for example from CME to GK to CCM ) like ip qos dscp cs3 signaling can one of these methods do the whole job of matching so
[OSL | CCIE_Voice] Proctor Labs is down
I have lost connectivity with proctor labs, my VPN client is connected but I cannot access via the web. I logged onto work to try a different ISP and that is the same. Looks like that is the end of my lab for today ... Trevor
Re: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config
I have had many replies and verification of my configuration, thank you all for that. Yes the VLANs were ceated and verified I could also ping the VLAN interfaces. So the scenario is this the interfaces and vlans are configured correctly as is the DHCP scope. The phones pick up an address from DHCP and then drop it, if I shut/no shut the interfaces they will pick up the DHCP ip again and then drop it. I cleared CDP else the IP would stay there. This has happened to me 2 days running I wonder if the module could be checked ? I definatley went over the 10 minute troubleshooting rule on this one From: Trevor Peddle [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Friday, 24 October, 2008 18:55:24 Subject: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config Hi all, I have configured the phone ports on BR1 as follows, they get an IP from the DHCP scope on CUCM server which I can see also with CDP on the router. However they are not contactable via IP, if I clear the CDP table the addresses disapear. I can ping the default gateway, it is correct in DHCP as is the net mask. I sometimes get a response via ping but probably a couple out of the blue now and again. The VLAN interfaces's are set correctly because I can see them ok. interface FastEthernet1/0 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 interface FastEthernet1/8 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 I configured switchport trunk encapsulation dot1q but it does not show in the config I also tried just as an access port with access vlan and voice vlan, also without success. This seems such an easy issue as I have never had such issues before when configuring similar ? I know my brain is swimming at the moment, first lab attempt on 6th Nov, but I do not think I have missed anything? I had the same problem on the same pod yesterday.
Re: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config
I just logged into pod 26 for my afternoon session. I will let you know if i have problems. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/ -Original Message- From: Trevor Peddle [EMAIL PROTECTED] Sent: Friday, October 24, 2008 4:16 PM To: ccie_voice@onlinestudylist.com ccie_voice@onlinestudylist.com Subject: Re: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config I have had many replies and verification of my configuration, thank you all for that. Yes the VLANs were ceated and verified I could also ping the VLAN interfaces. So the scenario is this the interfaces and vlans are configured correctly as is the DHCP scope. The phones pick up an address from DHCP and then drop it, if I shut/no shut the interfaces they will pick up the DHCP ip again and then drop it. I cleared CDP else the IP would stay there. This has happened to me 2 days running I wonder if the module could be checked ? I definatley went over the 10 minute troubleshooting rule on this one From: Trevor Peddle [EMAIL PROTECTED] To: ccie_voice@onlinestudylist.com Sent: Friday, 24 October, 2008 18:55:24 Subject: [OSL | CCIE_Voice] Pod 26 - BR1 NM-16ESW port config Hi all, I have configured the phone ports on BR1 as follows, they get an IP from the DHCP scope on CUCM server which I can see also with CDP on the router. However they are not contactable via IP, if I clear the CDP table the addresses disapear. I can ping the default gateway, it is correct in DHCP as is the net mask. I sometimes get a response via ping but probably a couple out of the blue now and again. The VLAN interfaces's are set correctly because I can see them ok. interface FastEthernet1/0 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 interface FastEthernet1/8 switchport trunk native vlan 360 switchport mode trunk switchport voice vlan 460 I configured switchport trunk encapsulation dot1q but it does not show in the config I also tried just as an access port with access vlan and voice vlan, also without success. This seems such an easy issue as I have never had such issues before when configuring similar ? I know my brain is swimming at the moment, first lab attempt on 6th Nov, but I do not think I have missed anything? I had the same problem on the same pod yesterday.
[OSL | CCIE_Voice] POD26 Subscriber not available
Proctor lab folks. POD26 subscriber is not accessible. I tried to open a ticket for after hours support but both links sends you to the support forum. I have posted the issue there as well. Scott. Scott Hardesty | Cisco Engineer | MidAtlantic | Presidio Networked Solutions 7601 Ora Glen Drive, Suite 100, Greenbelt, MD 20770 | mailto:[EMAIL PROTECTED] D: 301.313.2041 | C: 443.789.1219 | http://www.presidio.com/
[OSL | CCIE_Voice] MGCP Gateway POTS dial-peer
Hi, For Cisco IOS Software Release 12.3(7)T or later the Pots dial-peer configuration for MGCP gateway should like below or even service mgcpapp is not needed ? Could you please correct me if I am wrong ? dial-peer voice 10 pots service mgcpapp incoming called-number . direct-inward-dial port 1/0:15 Thanks, Bala. Get your preferred Email name! Now you can @ymail.com and @rocketmail.com. http://mail.promotions.yahoo.com/newdomains/aa/
