SIP ALG for NAT Cisco 3845
Does the SIP ALG for NAT have to be on the router where I am planning on running Call Manager Express for that to work properly, or can I have it running on an upstream router where I am currently preforming NAT functions?
You should be able to run it on either. If you have a choice, it would probably be better to run it on the upstream router. There are some things with SIP that aren't supported with the NAT ALG when the SIP is originated and fixed on the same router. One of these things I can name off hand is the 'sip bind' command which isn't supported on the same box.
It's still very possible to run it on the same box, however. But the SIP ALG doesn't require any information from CME to work properly.
Similar Messages
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SIP ALG / SIP NAT Traversal
I have a 2900 series router running IOS version 15.1(4). I am trying to connect 3rd party sip softphones to a 3rd party SIP Call controller on the inside. With low-cost firewall/gateways, I normally enable the SIP ALG feature and it will dynamically open the UDP ports for a SIP conversation for the duration of it and then close them. Does cisco IOS firewall have a SIP ALG feature and how do I configure it? Any guidance is much appreciated.
EddieIm trying to connect a SIP softphone (on the outside) to a IP PBX on the inside. I am seeing postings that say that "ip nat service sip" is the command that enables that feature, and others say that it breaks it. So far my testings shows that it does break it. Ultimately I want my outside softphone to register to the Phone system as an external IP address. It seems like SIP normally relays the internal IP address and the ALG router will make the translation on outbound and send it to the right source.
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I was wondering if anyone had a CUBE SIP Profile example for rewriting SDP to fix private-to-public IP address in the SDP so that CUBE can be used behind a static NAT without SIP ALG.
Im trying to connect a SIP softphone (on the outside) to a IP PBX on the inside. I am seeing postings that say that "ip nat service sip" is the command that enables that feature, and others say that it breaks it. So far my testings shows that it does break it. Ultimately I want my outside softphone to register to the Phone system as an external IP address. It seems like SIP normally relays the internal IP address and the ALG router will make the translation on outbound and send it to the right source.
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SIP over UDP routing in Cisco 3845
dear friends,
How can we configure the SIP over UDP protocol by Cisco 3845 router?
For more details please fine the attachment.Yes I tried but that is not helpful for me
How can I contact those people (engage a reputable consultant, or Cisco partner)?
Also I tried this commands in below.
voice class codec 1
codec preference 1 g711alaw
dial-peer voice 3250 voip
destination-pattern 3250
session protocol sipv2
session target ipv4:10.156.67.6
session transport udp
codec g711ulaw
sip-ua
retry invite 2
retry response 2
retry bye 2
retry cancel 2
no inband-alerting
sip-server ipv4:10.156.67.6
ip classless
ip route 0.0.0.0 0.0.0.0 10.157.67.1
ip route 10.157.67.0 255.255.255.0 10.167.67.225
access-list 101 permit ip host 10.156.67.1 host 10.156.67.100
access-list 101 deny udp any eq rip any
access-list 101 deny udp any any eq rip
access-list 101 deny udp any eq isakmp any
access-list 101 deny udp any any eq isakmp
access-list 101 permit ip any any
snmp-server engineID local 000000090200003094202740
snmp-server community public RW -
Configuring QoS on Cisco 3845 router for Polycom Video Conferecing
Dear All,
We have implemented a Polycom Video Conferecing solution at our Head Office. Using this we communicate with other branch offices through WAN (2mbps, MPLS).
The problem is that this WAN link is also used for data. When the traffic is high on the link, the voice and the video quality goes down drastically and we experience connection drops.
At the moment we have configured our Polycom box to communicate at 512kbps speed and we would like to reserve it in our WAN link. In case, video conferencing is not happening we would like it to be utilised by other traffic.
Can we configure QoS on our Cisco 3845 router to do this? I'm not a Cisco expert and have pressures from Management to correct this before the next conference.
I have already googled a fair bit but couldn't find something for me.
Could someone please tell me the exact commands that need to be given on our router to achieve this.
I'll be very thankful for this help.
Best Regards.Hi,
You can use something like the following to guarantee 5122k of bandwidth to your video-conferencing bandwidth but to allow that bandwidth to be used by other traffic when it is not being used for video-conferencing:
class-map VDOConf
match ip dscp af41
policy-map WANPolicy
class VDOConf
bandwidth 512
interface
service-policy output WANPolicy
Note that the above assumes that your video conferencing traffic is being marked to AF41. If that is not the case, you can always match on the IP address of your polycom device using an ACL:
class-map VDOConf
match access-group 101
access-list 101 permit ip .....
Hope that helps - pls rate the post if it does.
Paresh -
Cisco SCCP ATA or Linksys SIP ATA for hotdial
I need to install hotdial phones onto public access premises.
I think that the best choice for security reasons is to use simple analog phones connected to ATAs.
And now the question.
How can I configure Cisco SCCP ATA for hotdial? (I know how to configure IP phone for hotdial but this configuration don't work with SCCP ATA)
or
How can I register Linksys SIP ATA on a Cisco CallManager? (I know how to configure it for a hotdial)What version of sip are you using ?
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Does Cisco 3845 with NM-16A/S support OIR feature or Hot swap for this NM.
Dear Sir
My customer would like to implement Cisco 3845 with NM-16A/S x 4. I found that Cisco 3845 support OIR function but I am not sure OIR function that Cisco 3845 support, it support with which NM models. Can anyone tell me that NM-16A/S on Cisco 3845 support OIR function on this NM or not.
Thank you very much
WisitHi,
From what I have read from the following document.
http://www.cisco.com/en/US/products/ps5855/products_installation_guide_chapter09186a00802ccf1d.html
Network Modules
Network modules install directly into slots in the rear of the router. The Cisco 3845 router supports online insertion and removal (OIR, or hot swap) of network modules. The Cisco 3825 router does not support OIR.
Caution The Cisco 3845 router supports OIR with similar modules only. If you remove a network module, along with any installed WAN or voice interface cards, install another module and card combination exactly like it.
Interface Cards
Cisco 3800 series routers do not support OIR (hot swap) of interface cards inserted directly into router slots. You must turn off the router before installing or removing an interface card.
The Cisco 3825 router and the Cisco 3845 router each provide four interface card slots, labeled on the rear panel by HWIC and a number. Each slot can be occupied by one single-wide WIC, VIC, VWIC, or HWIC.
Hope this helps
Sarb -
Hi
Since 2 week I have some problem with my voip provider .
In my pabx software (3cx) my voip provider is correctly register but sometimes i dont know why I can't received external to internal call.
When i call my voip provider he tell me to disable sip alg of my router but in my RV016 i don't find any option sip alg.
How must i do?
My voip provider tell me to do the command line
no ip nat service sip udp port 5060
in telnet but i search and i can't acces telnet because i don't know login/password
Thanks for your help
Best Regard
LoicHi Loic,
Can you try please from the browser : https://IPaddress_of_rv016/f_general_hidden.htm
You will see the SIP ALG option
Please rate the post or mark as answered to help other Cisco Customers
thanks
Mehdi -
I have few queries related to SIP ALG feature.
- Is SIP ALG(NAT) and SIP inspection(inspect sip) dependent on each other? Do we need to enable both features on a router? Or are they independent?
My understanding is both are separate features and SIP ALG make sure it open up doors on NAT for any incoming connections from outside using the ports negotiated. And SIP inspection(the 'inspect sip' command) can be added for security purposes to make sure the router when acting as a firewall dynamically allows ports for traversal.
Please correct if my understanding is correct.
- Why do docs/discussions mention SIP ALG feature (enabled by default) breaks communication(one way/disconnects)? Can someone illustrate why would this feature result in a call breaking?
- Is there a difference in these two features in the way it is implemented on an ASA vs routers?
Would be grateful if someone could help on my queries. Tried searching in internet but didn't find clear info.
Thanks,
KarthicHello,
It has been always supported on the ASA platform. You just need to turn on the SIP inspection.
Mike -
Hi
I am looking to use a softphone from a third party software and for that I have just purchased a new BT Home Hub 3 in the hope that it would work (I had BT Home Hub 2 and was told to buy the Home Hub 3) but unfortunately it doesn't.
I was told that ONLY BT can disable my sip/alg and for that I phone the Technical support about 6 times and nobody seems to know what I am talking about. Now, I am with a second router and I cannot get what I am looking for, I am so frustrated.
Does anyone know how I could get the sip/alg disabled?
Many thanks,
KarineYou should be able to run it on either. If you have a choice, it would probably be better to run it on the upstream router. There are some things with SIP that aren't supported with the NAT ALG when the SIP is originated and fixed on the same router. One of these things I can name off hand is the 'sip bind' command which isn't supported on the same box.
It's still very possible to run it on the same box, however. But the SIP ALG doesn't require any information from CME to work properly. -
Hello;
I am attempting to assist a friend who was given a DDR2200 by their ISP, Centurytel.
We are trying to disable the SIP ALG in the gateway but the link is not present when we log into the gateways GUI.
We've tried to get Centurytel to assist us but all they have done is try to update the firmware.
software version v00.00.03.45.4e
hardware version v06
When we show them where the SIP ALG should be located in the GUI when referencing the DDR2200 user manual (pg 63) but they are not able to explain why we cannot see the link and are unable to assist us further.
I need to disable the SIP ALG as it's interfering with a SIP based Audio/Video Chat application my friend uses.
Any assistance would be greatly appreciated?
Thanks in advance.
Posted by WebUser Matthew WichlanHello.
The standard port for SIP is port 5060, so Cisco runs an ALG on this port.
You've switched it off, so there os NO ALG running for UDP at least.
Non standard ports won't be running the SIP ALG by default, although you can switch one on using the command you found.
Adam -
Can i disable the wrv210 sip alg ? it messes with my voip provider and must be disabled...i use a pap2T adapter and it works fine for outgoing calls...but after a while incoming calls half ring and i cant answer them.
please look at this link.....it doesn't state the info on the SIP ALG everyone you look at info on the WRV210...but this clearly states it has it built in to support voip to help traverse the firewall.....but it does it behind the scenes and cannot be turned off
http://www.cisco.com/en/US/prod/collateral/routers/ps9923/ps9929/data_sheet_c78-502735.html -
Problem with L2TP with Cisco 3845
Dear all
I have the following scenario for my dailup network.
MaxTNT(LAC) ---Ethernet--- Cisco3845 (LNS)
I have configuered MaxTNT Dailup server to act as LAC and launch a L2TP Tunnel after authenticating with Radius Server. Cisco 3845 acting as LNS estblishes L2TP tunnel with LAC and Dailup Users get connected on it as VPDNpppOE users.
However problem i am facing is that i don't receieve any authentication request on Cisco LNS. As soon as user gets connect it sents Accouting Request only.
I need authorization request in order to Push various different AVP from radius. But its not happening.
Anyone have any idea what could be wrong here?? is thre any specific parameter i need to set up Cisoc.. or on MaxTNT????
Waiting for replyTo enable the Layer 2 Tunnel Protocol (L2TP) tunnel server or network access server (NAS) to perform remote authentication, authorization, and accounting (AAA) tunnel authentication and authorization, use the vpdn tunnel authorization network command in global configuration mode. To disable remote tunnel authentication and authorization and return to the default of local tunnel authentication and authorization, use the no form of this command.
vpdn tunnel authorization network {list-name | default}
no vpdn tunnel authorization network {list-name | default} -
Hi all,
I'm having high CPU usage with one of my Cisco 3845.
It works as an IP-IP Gateway and the CPU is quite high when the total number of calls only around 100-200 calls.
I check the CPU usage with "show process cpu sort" and it looks like there are some "hidden" processes that consuming CPU.
For example, 41% is total CPU, 25% is due to interrups, so CPU utilization on process level = 41 - 25 = 16%.
But as showed below, processes don't consume that much CPU, only around 7% ???
Please help to advise on this case. Any help is highly appreciated..
Thank you.
3845-GW#show process cpu sort | ex 0.00% 0.00% 0.00%
CPU utilization for five seconds: 41%/25%; one minute: 46%; five minutes: 47%
PID Runtime(ms) Invoked uSecs 5Sec 1Min 5Min TTY Process
382 6619708 1473171 4493 1.59% 1.81% 1.92% 0 CCSIP_SPI_CONTRO
141 4228940 10181955 415 1.35% 1.51% 1.57% 0 IP Input
65 2450824 163102 15026 1.19% 1.16% 1.17% 0 Per-Second Jobs
370 2702292 3709512 728 0.87% 0.88% 0.88% 0 VOIP_RTCP
224 321680 245640 1309 0.47% 0.49% 0.50% 0 AFW_application_
112 93940 18093506 5 0.39% 0.31% 0.32% 0 Ethernet Msec Ti
384 1058280 1553567 681 0.23% 0.28% 0.30% 0 CCSIP_UDP_SOCKET
2 18148 32905 551 0.07% 0.03% 0.02% 0 Load Meter
137 35644 4657843 7 0.07% 0.04% 0.05% 0 IPAM Manager
189 206392 267959 770 0.07% 0.05% 0.07% 0 TCP Protocols
30 30792 198554 155 0.07% 0.01% 0.00% 0 ARP Input
368 145456 176151 825 0.07% 0.04% 0.05% 0 CC-API_VCM
28 9628 32759 293 0.00% 0.01% 0.00% 0 Environmental mo
48 221352 37922 5837 0.00% 0.11% 0.11% 0 Net Background
63 16728 32924 508 0.00% 0.01% 0.00% 0 Compute load avg
64 72080 2781 25918 0.00% 0.01% 0.00% 0 Per-minute Jobs
6 371644 29792 12474 0.00% 0.14% 0.12% 0 Check heaps
176 12216 240288 50 0.00% 0.01% 0.00% 0 CEF: IPv4 proces
284 36416 4929826 7 0.00% 0.02% 0.01% 0 MMON MENG
307 12168 806151 15 0.00% 0.01% 0.00% 0 Atheros LED Ctro
335 35300 19755 1786 0.00% 3.16% 1.00% 708 Virtual Exec
3845-GW#sh int g0/0
GigabitEthernet0/0 is up, line protocol is up
MTU 1500 bytes, BW 1000000 Kbit/sec, DLY 10 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation ARPA, loopback not set
Keepalive set (10 sec)
Full Duplex, 1Gbps, media type is RJ45
output flow-control is XON, input flow-control is XON
ARP type: ARPA, ARP Timeout 04:00:00
Last input 00:00:00, output 00:00:00, output hang never
Last clearing of "show interface" counters never
Input queue: 0/75/2/56803 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: fifo
Output queue: 0/40 (size/max)
5 minute input rate 1551000 bits/sec, 5751 packets/sec
5 minute output rate 4207000 bits/sec, 7643 packets/sec
925128804 packets input, 939078510 bytes, 0 no buffer
Received 62732 broadcasts (0 IP multicasts)
0 runts, 0 giants, 2 throttles
2 input errors, 0 CRC, 0 frame, 2 overrun, 0 ignored
0 watchdog, 3763438515 multicast, 0 pause input
1472816545 packets output, 3214770103 bytes, 0 underruns
0 output errors, 2067720191 collisions, 1 interface resets
0 unknown protocol drops
0 babbles, 2281155551 late collision, 0 deferred
2 lost carrier, 0 no carrier, 0 pause output
0 output buffer failures, 0 output buffers swapped out
3845-GW#sh int g0/1
GigabitEthernet0/1 is up, line protocol is up
MTU 1500 bytes, BW 1000000 Kbit/sec, DLY 10 usec,
reliability 255/255, txload 1/255, rxload 1/255
Encapsulation ARPA, loopback not set
Keepalive set (10 sec)
Full Duplex, 1Gbps, media type is RJ45
output flow-control is XON, input flow-control is XON
ARP type: ARPA, ARP Timeout 04:00:00
Last input 00:00:00, output 00:00:00, output hang never
Last clearing of "show interface" counters never
Input queue: 0/75/0/30335 (size/max/drops/flushes); Total output drops: 0
Queueing strategy: fifo
Output queue: 0/40 (size/max)
5 minute input rate 1684000 bits/sec, 7697 packets/sec
5 minute output rate 3372000 bits/sec, 5632 packets/sec
1484558664 packets input, 2383177786 bytes, 0 no buffer
Received 208998 broadcasts (0 IP multicasts)
0 runts, 0 giants, 0 throttles
2 input errors, 0 CRC, 0 frame, 2 overrun, 0 ignored
0 watchdog, 3060386282 multicast, 0 pause input
903478941 packets output, 2814588854 bytes, 0 underruns
0 output errors, 2910776303 collisions, 1 interface resets
0 unknown protocol drops
0 babbles, 4157448025 late collision, 0 deferred
2 lost carrier, 0 no carrier, 0 pause output
0 output buffer failures, 0 output buffers swapped outHas this been something that just recently started happening, or have you had this issue for a while? Have you installed any new programs recently?
You may want to download Glary Utilities, which is a free software(they will ask you if you want to go Pro, just say no, the free version works very well). There is a module for startup manager. You can go in and disable stuff that starts with the computer. I would advise unchecking adobe, java, quicktime, printers, etc. Anything that doesn't REALLY need to start with the computer. The nice thing with Glary is that you can restart the computer, and if you find that you need one of the programs to start with windows, you can go back in and enable it again.
The Celeron 925 processor in your computer is a decent entry level processor, but if there are too many programs running in the background, it can bog down quick. I would also recommend downloading and running Malwarebytes Anti-malware, to be sure that there is nothing malicous running in the background.
Qosmio X875 i7-3630QM, 32GB RAM, OCZ SSD Qosmio X505 i7-920XM, PM55, 16GB RAM, OCZ SSD
Satellite Pro L350 T9900, GM45, 8GB RAM , Intel 320 SSD (my baby) Satellite L655 i7-620M, HM55, 8GB RAM, Intel 710 SSD (travel system) -
Sip passing through nat but rtp is not - no audio
Sip passing through nat but rtp is not
I'm looking at traffic leaving my router with a sniffer. I see SIP traffic but I do not see RTP traffic. The phones ring on both sides but I do not get any audio.
interface f0/0.100
ip address 192.168.10.1 255.255.255.0
ip nat outside
ip nat pool VoIP 192.168.10.1 192.168.10.1 prefix-length 24
ip nat inside source route-map VoIP pool VoIP overload
ip nat inside source static tcp 10.1.1.2 49201 192.168.10.54 49201 extendable
access-list 1 permit ip host 10.1.1.2 any
route-map VoIP permit 10
match ip address 1
match interface f0/0.100
set interface f0/0.100Hello,
You can enable "ip nat service sip" or "ip nat service h323" and "ip nat
service h225" commands. As per the documentation, they are enabled by
default. In the latest IOS there is a new feature added to Cisco IOS that
ensures that even RTP packets get translated to one of the allowed ports as
specified by the RFC. The command to enable the feature is "ip nat service
allow-sip-even-rtp-ports"
http://www.cisco.com/en/US/prod/collateral/iosswrel/ps6537/ps6586/ps6640/pro
d_white_paper0900aecd80597bc7.html
Hope this helps.
Regards,
NT
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