VoIP+QoS+xDSL+H.323Gatekeeper [7:57104]
Well, this should give you enough to chew on since voice is becoming a hot topic. I am trying to configure VoIP with QoS. Why over IP and not over ATM, you say? I have to controll the call with a H.323 Gatekeeper, and that is IP. My problem appears to be that the call setup (or maybe signalling?) appears to be delayed. The test results are as follows: If the WAN link is saturated with data packets PRIOR to establishing the voice call, the first 10 to 15 (approximately) seconds of the call are lost. After the call is established, voice is rock solid and no voice packets are delayed or lost. If the voice call is established PRIOR to saturating the WAN link with data packets, the voice call is rock solid and no voice packets are delayed or lost. Thoughts or configs would be appreciated. --Mark version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname Router ! logging buffered 4096 debugging ! memory-size iomem 25 ip subnet-zero ! no ip domain lookup ! ip cef ! voice call carrier capacity active voice rtp send-recv ! no voice hpi capture buffer no voice hpi capture destination ! vc-class atm vip vbr-rt 256 256 10 precedence 5 no bump traffic no protect vc no protect group ! vc-class atm normal vbr-nrt 192 192 precedence other no protect vc no protect group ! interface ATM0/0 ip address 1.1.1.254 255.255.255.0 ip nat outside no atm ilmi-keepalive bundle-enable bundle qosmap protocol ip 1.1.1.1 encapsulation aal5snap pvc-bundle data 0/37 class-vc normal pvc-bundle voice 0/36 class-vc vip ! dsl equipment-type CPE dsl operating-mode GSHDSL symmetric annex A dsl linerate AUTO h323-gateway voip interface h323-gateway voip id Gatekeeper ipaddr x.x.x.x 1718 h323-gateway voip h323-id Gateway ip rsvp bandwidth 64 64 ip rsvp resource-provider wfq pvc ! interface FastEthernet0/0 ip address 10.200.100.1 255.255.255.0 ip nat inside speed auto ! ip nat inside source list 1 interface ATM0/0 overload ip classless ip route 0.0.0.0 0.0.0.0 1.1.1.1 no ip http server ip pim bidir-enable ! access-list 1 permit 10.200.100.0 0.0.0.255 ! call rsvp-sync ! voice-port 2/0 station-id name StaID station-id number 111222 caller-id enable ! voice-port 2/1 station-id name StaID station-id number 111222 caller-id enable ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern T session target ras ! gateway ! line con 0 line aux 0 line vty 0 4 login ! no scheduler allocate end Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57104&t=57104 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: how to configure WIC Fractional T1 with subint [7:57115]
What you have is a good start. The following is a sample of something I did years ago and it works with NAT. interface Serial0/0 no ip address no ip directed-broadcast encapsulation frame-relay no ip mroute-cache ! interface Serial0/0.102 multipoint ip address 192.168.1.21 255.255.255.252 no ip directed-broadcast ip nat inside frame-relay interface-dlci 102 ! interface Serial0/0.103 multipoint ip address 192.168.1.5 255.255.255.252 no ip directed-broadcast ip nat inside frame-relay interface-dlci 103 etc., etc., etc --Mark Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57117&t=57115 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
VoIP+QoS+xDSL+H.323Gatekeeper [7:57121]
For those of you trying to email me from the link in the message, here is the updated post. Sorry about the duplicate. *** Well, this should give you enough to chew on since voice is becoming a hot topic. I am trying to configure VoIP with QoS. Why over IP and not over ATM, you say? I have to controll the call with a H.323 Gatekeeper, and that is IP. My problem appears to be that the call setup (or maybe signalling?) appears to be delayed. The test results are as follows: If the WAN link is saturated with data packets PRIOR to establishing the voice call, the first 10 to 15 (approximately) seconds of the call are lost. After the call is established, voice is rock solid and no voice packets are delayed or lost. If the voice call is established PRIOR to saturating the WAN link with data packets, the voice call is rock solid and no voice packets are delayed or lost. Thoughts or configs would be appreciated. --Mark version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname Router ! logging buffered 4096 debugging ! memory-size iomem 25 ip subnet-zero ! no ip domain lookup ! ip cef ! voice call carrier capacity active voice rtp send-recv ! no voice hpi capture buffer no voice hpi capture destination ! vc-class atm vip vbr-rt 256 256 10 precedence 5 no bump traffic no protect vc no protect group ! vc-class atm normal vbr-nrt 192 192 precedence other no protect vc no protect group ! interface ATM0/0 ip address 1.1.1.254 255.255.255.0 ip nat outside no atm ilmi-keepalive bundle-enable bundle qosmap protocol ip 1.1.1.1 encapsulation aal5snap pvc-bundle data 0/37 class-vc normal pvc-bundle voice 0/36 class-vc vip ! dsl equipment-type CPE dsl operating-mode GSHDSL symmetric annex A dsl linerate AUTO h323-gateway voip interface h323-gateway voip id Gatekeeper ipaddr x.x.x.x 1718 h323-gateway voip h323-id Gateway ip rsvp bandwidth 64 64 ip rsvp resource-provider wfq pvc ! interface FastEthernet0/0 ip address 10.200.100.1 255.255.255.0 ip nat inside speed auto ! ip nat inside source list 1 interface ATM0/0 overload ip classless ip route 0.0.0.0 0.0.0.0 1.1.1.1 no ip http server ip pim bidir-enable ! access-list 1 permit 10.200.100.0 0.0.0.255 ! call rsvp-sync ! voice-port 2/0 station-id name StaID station-id number 111222 caller-id enable ! voice-port 2/1 station-id name StaID station-id number 111222 caller-id enable ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern T session target ras ! gateway ! line con 0 line aux 0 line vty 0 4 login ! no scheduler allocate end Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57121&t=57121 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: VoIP+QoS+xDSL+H.323Gatekeeper [7:57121]
That is correct--I hear silence when the data pvc is saturated. However, further tests performed just recently indicate that the amount of data saturating the link corresponds to the amount of lost voice packets. That is why I thought this was a QoS issue. It almost appears that some of the call control packets are going down the data pvc instead of the voice pvc. But I don't want to comit to such a statement for fear of biasing other opinions in the newsgroup, as others may have the "real" cause to the problem already figured out. I did explore an access-list config to match on port 1720 and there were some hits, but again are there other voice payload and/or voice signaling packets traversing the data pvc? I don't know. --Mark Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57133&t=57121 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP+QoS+xDSL+H.323Gatekeeper [7:57121]
Yeah, sorry about that. I added the following: ip qos dscp cs5 media ip qos dscp cs5 signaling to my dial-peer after the original post. Unfortunately, same result. Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57141&t=57121 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: VoIP+QoS+xDSL+H.323Gatekeeper [7:57121]
Hamid, Well there are two different ways that you can look at this. From the telco side one may conclude that the entire duration of the call is the actual cost of the call, including call setup. From the customer side, the majority of customers would argue that call setup should not be billed, as that is not part of the actual voice conversation. However, ultimately, all costs are eventually passed to the consumer. Unfortunately, this doesn't directly answer your question but does explain the two theories. I don't get into call accounting too much. --Mark Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57144&t=57121 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Re: VoIP+QoS+xDSL+H.323Gatekeeper [7:57121]
To paint you a better picture here is the scenerio...I am calling a number that I know will respond initially with a voice annoincement (ie. Voice mail system). When data is not flooding the wan link the call is fine. I get the entire announcement beginning with "Thank you for calling xyz company, the nations leading provider of..." When data IS flooding the wan link, the call eventually goes through but I hear the recording starting in the middle of the stream... "leading provider of..." In the second case, why wouldn't I hear the entire message, just delayed until the call set-up packets traversed the network? ""Mark S"" wrote in message news:200211081804.SAA07463@;groupstudy.com... For those of you trying to email me from the link in the message, here is the updated post. Sorry about the duplicate. *** Well, this should give you enough to chew on since voice is becoming a hot topic. I am trying to configure VoIP with QoS. Why over IP and not over ATM, you say? I have to controll the call with a H.323 Gatekeeper, and that is IP. My problem appears to be that the call setup (or maybe signalling?) appears to be delayed. The test results are as follows: If the WAN link is saturated with data packets PRIOR to establishing the voice call, the first 10 to 15 (approximately) seconds of the call are lost. After the call is established, voice is rock solid and no voice packets are delayed or lost. If the voice call is established PRIOR to saturating the WAN link with data packets, the voice call is rock solid and no voice packets are delayed or lost. Thoughts or configs would be appreciated. --Mark version 12.2 service timestamps debug datetime msec service timestamps log datetime msec no service password-encryption ! hostname Router ! logging buffered 4096 debugging ! memory-size iomem 25 ip subnet-zero ! no ip domain lookup ! ip cef ! voice call carrier capacity active voice rtp send-recv ! no voice hpi capture buffer no voice hpi capture destination ! vc-class atm vip vbr-rt 256 256 10 precedence 5 no bump traffic no protect vc no protect group ! vc-class atm normal vbr-nrt 192 192 precedence other no protect vc no protect group ! interface ATM0/0 ip address 1.1.1.254 255.255.255.0 ip nat outside no atm ilmi-keepalive bundle-enable bundle qosmap protocol ip 1.1.1.1 encapsulation aal5snap pvc-bundle data 0/37 class-vc normal pvc-bundle voice 0/36 class-vc vip ! dsl equipment-type CPE dsl operating-mode GSHDSL symmetric annex A dsl linerate AUTO h323-gateway voip interface h323-gateway voip id Gatekeeper ipaddr x.x.x.x 1718 h323-gateway voip h323-id Gateway ip rsvp bandwidth 64 64 ip rsvp resource-provider wfq pvc ! interface FastEthernet0/0 ip address 10.200.100.1 255.255.255.0 ip nat inside speed auto ! ip nat inside source list 1 interface ATM0/0 overload ip classless ip route 0.0.0.0 0.0.0.0 1.1.1.1 no ip http server ip pim bidir-enable ! access-list 1 permit 10.200.100.0 0.0.0.255 ! call rsvp-sync ! voice-port 2/0 station-id name StaID station-id number 111222 caller-id enable ! voice-port 2/1 station-id name StaID station-id number 111222 caller-id enable ! dial-peer cor custom ! dial-peer voice 1 voip destination-pattern T session target ras ! gateway ! line con 0 line aux 0 line vty 0 4 login ! no scheduler allocate end Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57351&t=57121 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
Packets received on wrong pvc [7:57509]
I turned on some debugging and I am seeing this message in my log files. Does anyone know the cause or how to correct? IP precedence 0 received on voice but should have been on data --Mark Snips of the router config: class-map match-all voip match access-group 100 policy-map llq class voip priority 80 set ip dscp cs5 class class-default fair-queue set ip dscp default vc-class atm vip vbr-rt 256 256 20 precedence 5-7 no bump traffic no protect vc no protect group vc-class atm normal vbr-nrt 192 192 precedence 0-4 no protect vc no protect group end interface ATM0/0 bandwidth 256 ip address 1.1.1.254 255.255.255.0 no atm ilmi-keepalive bundle-enable bundle qosmap protocol ip 1.1.1.1 encapsulation aal5snap pvc-bundle voice 0/36 class-vc vip service-policy output llq pvc-bundle data 0/37 class-vc normal service-policy output llq ! dsl equipment-type CPE dsl operating-mode GSHDSL symmetric annex A dsl linerate AUTO access-list 100 permit ip any any dscp cs5 access-list 100 permit ip any any precedence critical access-list 100 permit tcp any any eq 1720 access-list 100 permit udp any any range 16383 16384 dial-peer voice 1 voip destination-pattern T session target ras ip qos dscp cs5 media ip qos dscp cs5 signaling Router#sh debug Generic ATM: ATM VC Bundle Events debugging is on ATM VC Bundle Errors debugging is on ATM VC Bundle INARP Events debugging is on ATM VC Bundle Adjacency Events debugging is on Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57509&t=57509 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
RE: Port Address Translation [7:57485]
HP uses ports starting at 9100 for their JetDirect product. If you are
trying to translate through a NAT router, try the following:
ip nat inside source static tcp {inside IP} 9100 {outside IP} 9100 extendable
--Mark
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Re: full duplex or half duplex, how can you tell [7:57431]
Wow! Two ethernets! Bonus! I once worked on a 2501 and I only had one ethernet...AND it was AUI! --Mark Message Posted at: http://www.groupstudy.com/form/read.php?f=7&i=57519&t=57431 -- FAQ, list archives, and subscription info: http://www.groupstudy.com/list/cisco.html Report misconduct and Nondisclosure violations to [EMAIL PROTECTED]
