Support Center

Quality of Service (QoS) & Router Guidelines

Last Updated: Nov 02, 2016 05:08AM EDT
In order for your network to work properly with Sharpen’s service, some changes may need to be made to devices and equipment. Please ensure the following conditions are met on any routers, firewalls, or modem/router combinations that are part of your network. These changes can typically be made in the device’s web interface.


Use Quality of Service to maintain prioritization.
Many devices support Quality of Service (QoS) tags to maintain traffic priority across the network. It may be beneficial to tag your voice traffic with the appropriate tags, so it can be prioritized anywhere in the network in the event of a saturated link. This will help to prevent any audio issues caused by voice and data competing for the same bandwidth over your Internet connection.

Use traffic shaping to offer voice traffic the necessary bandwidth.
Due to the relatively large amounts of data that voice uses on the network, it is important to ensure that your voice traffic has enough bandwidth to operate. As such, traffic shaping rules can be implemented to allow voice traffic to use additional bandwidth, or even limit other types of traffic to help out voice.

The default QoS settings are:
  • UDP/10000-20000 - Priority: High
  • UDP/5060-5081 - Priority: High
  • TCP/5060-5081 - Priority: High
Please note the following: 
  • Not all routers have QoS capabilities.
  • Some routers / firewalls have NAT / UDP timeouts for ports that the phones communicate on. Sharpen Service requires the timeouts for the specified ports above to be longer then 240 seconds(phones register every 240 seconds). 
The information below explains how are phones stay connected and pass traffic on a local network:
 SIP has two pieces to it.
1.) 5060 UDP traffic, this is the information about the call IE: Phone 1 calling Phone 2 via server xyz
2.) 10000-20000 UDP Traffic - This is the RTP Stream it hold the actual packets of voice data that make up the phone call.
While at rest the phones only send 5060 data as a Keep Alive method for NAT, during this period there is no RTP traffic.  However once a phone call is made and audio established RTP traffic is sent from the phone to our servers.

Router Guidelines

Double-NATing (Double-Routing): Ideally, you will need to have only one device performing routing functions. Double-NATing (double-routing) is known to cause many problems for VoIP phones. It is best to eliminate or bridge any extra or additional routers or modem/router combinations on your network. If you need to put your modem/router combination in bridge mode, please contact your internet service provider (ISP) for assistance. You may also try putting the second router in the first router’s DMZ if bridging is not possible, but this method is not guaranteed to work.

Allow SIP, UDP, and RTP Protocols: Sharpen uses these protocols to send and receive traffic. If you are using a firewall to secure any of these protocols, it will interfere with Sharpen traffic and may inhibit your ability to make or receive calls.

Because Sharpen's IP range is dynamic, we recommend the phones be allowed open access through your firewall. If you need additional information, please contact our support team at 855.249.3357 option 1 or at

Disable SPI (Stateful Packet Inspection): SPI allows the router to approve or deny any information packets that flow through it for security reasons. However, it often incorrectly identifies our VoIP traffic as a security risk. Disabling SPI will prevent this.

Disable SIP ALG (SIP Application Level Gateway) and/or SIP Transformations:  These are other security features that sometimes prevent our traffic from flowing properly. On Cisco routers, this is usually referred to as SIP inspection and can be disabled with the command 'No inspect SIP' or 'No Fixup Protocol SIP 5060'.

Disable any VoIP-specific functions: Networking equipment will often come customized for VoIP, but in many of these cases the customization interferes with our traffic. Ultimately, our system does not require specific VoIP-supporting functions.

Segregate voice traffic to its own VLAN.

Voice traffic tends to come in large amounts of two-way UDP communication. Since there is no overhead on UDP traffic ensuring delivery, voice traffic is extremely susceptible to bandwidth limitations, clogged links, or even just non-voice traffic on the same line. Separating out your voice traffic allows it to function independently of other network traffic, and allows for more granular control over different types of traffic.

Ensuring these recommendations  are in place should resolve most of your issues. After you have made the changes, you will need to restart your network. For further information on networking, please see the network guidelines article. 


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