VoIP Routers and QoS Primer

VoIP Quality of Service (QoS)

Update: Check out our VoIP router compatibility guide to help you quickly assess your problem.

You are sitting at your office desk, and suddenly you realize that your phone is not working or the voice quality is very non-human, almost like a robot! Adding further to your embarrassment all your incoming calls is going to your voicemail, and you don’t know what could be the reason! VoIP is such an unpredictable world, and usually, in a home and small office environment, such issues are commonplace. The main thing is to dive deep into some of these problems and to discuss which methods we can use to avoid such issues from happening.

Let’s start with a few essential issues most people usually face while using VoIP services via SOHO routers. Most of the routers are not quite safe while handling VoIP, thus causing one-way audio, phone registration, and connectivity loss issues. The most important features that any VoIP router needs is support for VoIP QoS, bandwidth prioritization for VoIP and SIP ALG.

VoIP is a complex technology which works by utilizing different protocols and services. A lot depends on exactly what you want out of it and how you want to make outbound calls and receive incoming calls. There are solutions like EdgeMarc Series routers which provide a VoIP aware NAT/Firewall, VoIP survivability, passive call quality monitoring and powerful but easy-to-use traffic management that ensures high-quality voice and video. At the same time, some routers are low in price but not very good in performance, and their QoS is marginal.

Some of the functions incorporated in different routers, which might interfere with VoIP and its QoS are:

  • ALG settings
  • SPI ( SIP Packet Inspection)
  • IDS/IPS ( Intrusion Detection System and Intrusion Prevention system)
  • NAT settings

Above mentioned settings can cause one way audio issues, poor voice quality issues, NAT traversal issues, problems in making outbound calls, difficulties in receiving calls usually because of the wrong implementation of the ALG on most of the routers. So it’s always good to disable these options on the router if faced any such issue. It’s also recommended to fine tune the NAT settings, so there is no problem NAT{ing} the incoming and outgoing packets on the router.

The problem with SIP ALG is that it modifies the SIP headers and SDP incorrectly. The simple logic used by SIP ALG is to replace any private IP address it finds in the outgoing SIP packets from the router public interface hence resulting in corrupting the SIP packets, writing incorrect information in the SIP header, changing the port number values, etc. which results in breaking the SIP call and creates problems in communication. To find a list of SIP ALG enabled routers, please visit this link on VoIP-Info portal: http://www.voip-info.org/wiki/view/Routers+SIP+ALG.

Sometimes VoIP becomes an issue in a corporate office environment if your bandwidth is being utilized by many users, who might be downloading different torrents and files while you will be trying hard to make some VoIP calls. Usually, it’s best to keep the internet for VoIP telephony separate from the commonly used internet, especially by creating separate VLANs for VoIP traffic.

VoIP QoS is not handled very well by most of the routers. Usually, no high bandwidth is provided to the incoming VoIP traffic, no prioritization of the packets is done because of which VoIP issues arises. VoIP QoS is usually implemented at the ISP core routers- all bets are off when it hits the first ISP router unless they are providing you VoIP services too, and then they often will prioritize VoIP traffic on their network as well. So it’s always best to first discuss with your Internet Service provider regarding their VoIP support and if any VoIP QoS prioritization is performed on their routers.

In most of the cases, we also need to check the firewall settings also. This firewall could be a built-in firewall in our Linux machine or some sophisticated one offered by CISCO or Juniper. An excellent guide to implementing SIP, MGCP, H.323, SCCP implementation on Cisco PIX/ASA 7.x can be found on: http://www.cisco.com/c/en/us/support/docs/security/asa-5500-x-series-next-generation-firewalls/82446-enable-voip-config.html. On Linux firewall, it’s best to keep the SIP and RTP ports open or you will face one-way audio and phone registration issues. An excellent primer on Linux firewall implementation for SIP can be found at http://www.voip-info.org/wiki/view/Asterisk+firewall+rules.

It’s a good design practice to deploy some network management tools in your corporate environment to monitor the inbound and outbound bandwidth on your network interfaces. There are many available, some open source tools like Nagios (http://www.nagios.org/), MRTG (http://oss.oetiker.ch/mrtg/) and SmokPing (http://oss.oetiker.ch/smokeping/) can be used to monitor the network performance in real time and make decisions on the base of the statistics.

VoIP Quality of Service (QoS)

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