Voice over IP via 2 netvanta 3448 issues

[dlazure]

I have a customer where we installed two netvanta 3448 to connect two site together. they use the link to connect two IP phone on the remote site and share files.

they complain about the voice being cut all the time. I programmed QoS but i am not sure if it is done correctly

can i get some help please

i joined both config

[Anonymous]

:

Thank you for asking this question in the support community.  When you get a chance, can you provide some additional information about the nature of the application?  Are the phones from the remote site registering to the phone system at the main site?  Are the phones going over the VPN?  Are the phones and the phone system using SIP?  You mentioned the phones will work for a while, and then what happens?  Do they drop in the middle of the call, does the audio become choppy, or something else?

Again, thanks for posting this question, and let me know some additional information, and I'll be happy to help in any way I can.

Levi

[dlazure]

? Are the phones from the remote site registering to the phone system at the main site? 

Yes

Are the phones going over the VPN?

Yes

Are the phones and the phone system using SIP? 

Yes

You mentioned the phones will work for a while, and then what happens?

they work all the time, but when you speak to someone the voice quality is bad and some audio is being cut and sometime the call drop

Do they drop in the middle of the call, does the audio become choppy, or something else?

The call drop or become choppy

[Anonymous]

You've got a few issues to resolve.

First remember that this is the Internet, that means no QoS no matter how you tag your traffic. I have seen ISPs offer QoS within their own core, but it would have to be something they are specifically providing.

Next, traffic shaping. Unless both directions have a 100Mbps synchronous connection to the Internet you want to be sure the router is aware that port speed is not the available bandwidth.

Now the big one. You are prioritizing outbound traffic to the Internet based on the UTP ports. BUT, those UTP packets are being encapsulated within the VPN packets which are TCP and from the outside they all look the same regardless of the traffic type/port. Also, encryption in general can cause quality issues because of increased latency. So I've worked around this a couple ways. You can do two VPN tunnels and prioritize the voice tunnel. This requires multiple IPs and really isn't my preferred. Another solution is to use the VPN only for data. Enable the ALG for SIP traffic, use static translations, and absolutely set up ACLs for inbound SIP traffic or you'll be hacked and calling other countries in no time.

Yes, this means you VoIP communications is unencrypted across the internet. But as long as you use the ACLs and your security is set up correctly the risk is low that someone would be able to intercept the traffic in a way that the participants wouldn't notice (such as one-way audio).

A variant of the two VPN tunnels is a Data VPN and a VoIP GRE. Prioritize the GRE, remember it's encapsulated. A little more secure but no encryption.

[dlazure]

Thanks rcohen for the reply

I want to program the qos by matching ip precedence

(config)#qos map VOIP 10

(config-qos-map)#match precedence 5

(config-qos-map)#bandwidth 640

my phone system is taging RTP traffic with a priority of 5

Can't i just apply this qos map to the vpn?

i know in cisco we can take the ip precedence value on the packet and place it outside the IPSEC payload during encryption.

Is there a way in AOS ?

thanks

[Anonymous]

That is a good question.

I haven't heard of AOS being able to tag VPN packets with IP Precedence to match the payload, but maybe I missed it, I'll leave this to Adtran to answer. Adtran, please post the answer as I too am very interested to know.

I do know that ISPs commonly strip QoS tags in transport, but you can't prioritize incoming traffic anyway.

AOS is a second language to me, IOS being my first. There is a lot you can do with IOS that AOS can't, but you get to pay a LOT more for it. For your basic one or two T1s, or <=75Mbps routing it's a great router. It also works very well with VoIP traffic, and is extremely reliable, but sometimes you need more. BTW when I talk about Cisco I mean the real IOS based Cisco, not that web-only, linksys-like crap they put the Cisco label on.

[Anonymous · Accepted solution]

:

Please, review the following post on VPN & QoS:  https://supportforums.adtran.com/message/1731#1731

Since you are going over the Internet, there is some QoS limitation because you have no control as to how traffic is prioritized over the connection. Although it is not as clear-cut as setting up QoS over a private network, we can set up a QoS policy to at least ensure that the NetVantas prioritize certain traffic types. When setting up QoS for voice traffic going over a VPN, you must match either the DSCP value or IP precedence value. Matching by any other parameter will not work as those other parameters will be encrypted before QoS has a chance to prioritize how the traffic is sent out.

Therefore, the QoS map you suggested is accurate, however, since the voice traffic is low latency, I would recommend the priority command instead of the bandwidth command, as follows:

(config)# qos map VOIP 10

(config-qos-map)# match precedence 5

(config-qos-map)# priority 640

I hope that makes sense, but let me know if you have any further questions.

Levi

[dlazure]

Thanks Levi for the reply

I have some other questions

Do I need to tag the RTP and SIP value with different priority?

How much bandwidth SIP need per phone call?

Can I log remotely to the 3448 and disable the actual qos map and aply the new one without droping the interface?

[Anonymous]

:

The configuration files you submitted previously were for NetVanta 3120s.  Are the units you have questions on NetVanta 3120s or NetVanta 3448s?  What is the type of WAN/Internet interface that the QoS map is applied to?  Applying a QoS Map to a PPP or demand interface will cause the interface to drop briefly.  This causes a temporary service interruption and the interface should come back up momentarily.

The amount of bandwidth usage per call, depends on the CODEC.  Typically G.711 uses approximately 90 Kbps per call, and G.729 uses about 40 Kbps per call.

Most likely, for a QoS over VPN application, the phones will mark their traffic with a DSCP or IP Precedence value, and that is what you will need to prioritize in the QoS map.  For additional information on QoS, the links below go over how to configure a basic QoS map over the web interface and CLI:

Configuring QoS in AOS in the GUI (video) - https://supportforums.adtran.com/videos/1046

Configuring QoS for Voice over PPP WAN Interface (CLI) - https://supportforums.adtran.com/docs/DOC-2654

Configuring QoS for Voice over Ethernet WAN Interface (CLI) - https://supportforums.adtran.com/docs/DOC-3406

Please, let me know what additional questions you have.

Levi

[dlazure]

We changed the 3120 for 3448 because adtran told me 3120 cpu were to weak to manage voip.

I applied my 3130 config over the 3448

So i am using two 3448 now

I dont understand one thing

You seem to include the sip and rtp bandwidth in the required bandwidth

I want to know i lf i need to tag the sip packet and rtp with different priority value ?

Dominic lazure

Luminet solutions

Message was edited by: levi (Removed Contact Information)

[Anonymous]

Dominic:

Most likely, for a QoS over VPN application, the phones will mark their traffic with a DSCP or IP Precedence value, and that is what you will need to match and prioritize in the QoS map (because those are the only values that can be matched and prioritized over a VPN).

Please, let me know what additional questions you have on this application.

Levi

[dlazure]

I understand but I want to know if the  sip portion need to be tagged a different value than the RTP.

[Anonymous]

Dominic:

It depends on the end equipment if SIP and RTP have different DSCP/IP Prec. values or if they are the same.  It doesn't matter to the ADTRAN unit if they are different or the same, but for QoS, if the SIP and RTP have different values, it is imperative that you at least match the RTP value.

I hope that answers your question, but if not, let me know.

Levi

[dlazure]

I removed all the qos map programmed to creat a new one and I lost my remote connection to the router and I need them to make a reset so I can login again

Why?

[Anonymous]

Dominic:

Unfortunately, the QoS map configuration does not control administrative access.  Administrative access is controlled by the firewall.  If you would like to reply with the changes you made, I will be happy to review them for you.

Levi

[dlazure]

Thanks Levi

All I did is by the web interface I went to Qos map and I selected all the maps and clicked on removed selected maps.

The customer lost his internet connectivity and I needed to reset the device to run on the startup config with the qos map enabled.

I don’t understand why it does that?

[Anonymous]

Dominic:

Changing the QoS maps, should not have anything to do with the ADTRAN's firewall administrative access.  At this point, I would suggest opening up a ticket with ADTRAN Technical Support to troubleshoot this issue (then you can post the results back here to this forum thread). Be sure to mention this forum post when you open the ticket. You can open a ticket in several ways:

- By web - Create a Service Request

- By emailing support@adtran.com

- By calling 1-888-423-8726

Levi

[dlazure]

Levi

I did some test and realized the 3448 was freezing if i removed the qos map insite to outside... im not sure if its a default map. I created the qos map VOIP

who match ip precedence 5 and 3 with a bandwitdh priority of 640 kb/s and applied it to my eth 0/1 interface and vlan 1. on both site

i can see the qos map matching a lot of packets so far. i will test next week with the customer

[dlazure]

Hi Levi

I applied the qosmap VOIP 10 to my eth 0/1 and vlan 1 and i still got issues...

I applied the map to ETH 0/1 OUT only as i was denied to apply it to INBOUND traffic because i use priority...

i think INBOUND traffic should also prioritize voip packets.. am i wrong?

i get this message : Can not set a QoS map configured with bandwidth requirements as the input qos-policy on an interface.

[Anonymous]

:

If you would like to reply with a copy of the current configuration (please, remember to remove any information that may be sensitive to the organization).  I will be happy to review it for you.

Levi