SIP priority/ QoS for Cisco VOIP phones question

Currently at my company we have three locations with Cisco VOIP phones running on an Asterisk system. Referring to the diagram below, the Asterisk server sits at the home location and all phones communicate back to it. From time to time I'm alerted to call quality issues from users at the two remote locations. What I've been tasked to do is ensure some kind of priority/QoS for phone traffic to eliminate these issues.

My question is, what's the best approach to tackle this? I've done some research and found articles on things like policy mapping to prioritize SIP traffic and an article on Auto QoS that involves Cisco vOIP phones. Overall I'm not sure exactly what I need.

Below is the network diagram with all of the devices involved, their license levels, the links between facilites and their speeds.

(Location 3) 1841 Router-adv ip services(Gateway)>-->3750 Switch-ipservices >--40 MBps MPLS--> (Location 2)1921 Router-ipbase >-->3750 Switch-ipservices (Gateway)>--100 MBps Fiber--> (Home Location) 3750 Switch-ipservices (Gateway)

Any help is appreciated.
travisryanAsked:
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Tony BalanquitCommented:
Cisco 1841 and 1921 ISR's currently does not support SIP. Requisites on implementing such is going with 2901 or above models of Cisco ISR. you can the link below.
ISR Comparison:
http://www.cisco.com/c/en/us/products/routers/2900-series-integrated-services-routers-isr/series-comparison.html

Hope this helps!
travisryanAuthor Commented:
I'm confused on what your comment means. The phones are have been working for a while, right now I'm just looking at the different ways I can prioritize traffic / implement some sort of QoS for calls.
Tony BalanquitCommented:
Hi travisryan, In order for the voice quality to go inrease. The best option would be as you mentioned configure a QOS on the 3750 switch that you have connecting to the VOIP lines. This would prioritize voice over the LAN connection and would help in the quality if the calls please refer to the link for some configuration guide on QOS.

http://www.cisco.com/c/en/us/support/docs/switches/catalyst-3750-series-switches/91862-cat3750-qos-config.html

Hope this helps!
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travisryanAuthor Commented:
Tony, thanks for the reply, I'll go over that link. In the mean time, I found this page: Auto QoS

From that page it looks like 3 different command groups and I could be done:
auto qos voip cisco-phone for the port the phone is on
auto qos voip trust for layer 2 uplinks
auto qos voip trust for layer 3 uplinks

Is it really that easy?
travisryanAuthor Commented:
Something else to mention, all of my phones are on a separate phone VLAN. I'm still looking over your link, Tony, but I wanted to bring that up in case it made things easier.
José MéndezCommented:
Prioritizing only SIP traffic is not the correct approach. SIP does not carry the voice of the users. Instead, you want to focus on treating UDP RTP traffic as high priority.

ALso, depending how your Asterisk is configured, if it is acting as media proxy, then you want to focus on how the network treats the RTP Audio traffic from the phones to Asterisk (its IP), cuz all the audio will go and traverse the asterisk server to reach the other phone and viceversa. YOu may set up something different with the

canreinvite=yes

parameter, which causes asterisk to negotiated the audio between the 2 phones. In that case, focus on prioritizing the quality of traffic between ip phone to ip phone.

Try using lower bitrate codecs such as G.729 or GSM, AND, very important, set a limit to the maximum amount of calls crossing the WAN link, the more calls you have, the more RTP Connections you have competing for the bandwidth.

You probably want to test how many calls you can establish simultaneously  to a particular site. If you have 3 connections with good quality, and then you connect a4th call, and the audio of all 4 starts degrading, that is aclear symptom that you should not allow more than 3 connections through the WAN link in that site. At that point, QoS won't help at all. You can't have 4 higly prioritized connections at the same time competing for the same pipe and expect to have good quality.

Hope that helps.

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travisryanAuthor Commented:
Willy, thanks for the thorough answer. After going over it with the Asterisk admin I'd like to focus on prioritizing that traffic instead of reserving a specified amount of bandwidth. We have a 100 Mbps pipe and a 40 Mbps so I don't think bandwidth is the issue.

Doing some more research I think I can just prioritize the VOIP vlan all of the phones are on. I still need to research more to figure out what specific commands I need.
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