FAQ – VoIP

PROBLEM #1: The loss of incoming calls.

If the Office phone device, such as IP Phone or IAD (aka ATA), is unable to register with myPBX service, then the ability to receive an incoming call on it is gone. If you have voicemail enabled on myPBX extension user, then failing to get the Office phone device to respond will route the call to your voicemail.  If you do not have voicemail enabled, there will be no where to route the call to and after a number of rings the call will fail and the caller will either hear a fast busy or a message that the call was unable to be placed as dialed.

Look to one of the following as most probable:

REASONS:

  • Loss of your Internet connection.
  • Loss of power.
  • The blocking of signaling to the Office phone device, either by a firewall or a failure of a router to translate NAT correctly.

FIX:

If you lost your Internet connection, try power cycling the modem, router and Office phone device.  If this fails, call your ISP or your IT Administrator.

NOTE: myPBX service has feature of “Call Forwarding” and “Follow Me”.  If this feature is enabled, then in the event of the loss of your Internet connection the call will automatically forward or followed to a preset number.

PROBLEM #2: My calls are choppy.

Choppy audio is usually caused by lack of adequate bandwidth or from Internet congestion slowing down the connection (latency).  If this is an ongoing problem check the following:

REASONS:

  • That a computer application (such as file sharing or on-line gaming) is not taking the bandwidth away from the VoIP connection.  Check for malware on your PC running an application without your knowledge.
  • Check your available bandwidth by doing a Bandwidth test.
  • Check for unusual latency or packet loss on your connection.VoIP delay or latency is characterized as the amount of time it takes for speech to exit the speaker’s mouth and reach the listener’s ear. Latency sounds like an echo. There are 3 types of delay commonly found in today’s VoIP networks;
    1. Propagation Delay: Light travels through a vacuum at a speed of 186,000 miles per second, and electrons travel through copper or fiber at approximately 125, 000 miles per second. A fiber network stretching halfway around the world (13, 000 miles) induces a one-way delay of about 70 milliseconds (70 ms). Although this delay is almost imperceptible to the human ear, propagation delays in conjunction with handling delays can cause noticeable speech degradation.
    2. Handling Delay: Devices that forward the frame through the network cause handling delay. Handling delays can impact traditional phone networks, but these delays are a larger issue in packetized environments.
    3. Queuing Delay: When packets are held in a queue because of congestion on an outbound interface, the result is queuing delay. Queuing delay occurs when more packets are sent out than the interface can handle at a given interval.
  • Check for Jitter, which is a common problem of the connectionless networks or packet switched networks. Because the information (voice packets) is divided into packets, each packet can travel by a different path from the sender to the receiver. When packets arrive at their intended destination in a different order then they were originally sent, the result is a call with poor or scrambled audio. Jitter is technically the measure of the variability over time of the latency across a network. Jitter is one of the most common VoIP call quality problems.
  • Inadequate Router
  • Internal Network Improperly Configured

FIX:

  • Check with your ISP and Network Administrator for a higher level of service with more bandwidth.  If you find high levels of latency or packet loss, let the ISP and Network Administrator check your connection.
  • Install a Specialized VoIP Router and Switch. Without a router and switch that is configured for VOIP packet prioritization, call quality can be impacted by the other users on your network. For example, if during a call, another user on your network downloads a large file, without packet prioritization, your call quality could be degraded. A VoIP router and switch prevents this from happening by giving priority to voice traffic on your network. VoIP routers are not an expensive piece of hardware these days.
  • Sometimes the adaptive Jitter Buffer buffer does not play nice.  So in your IP phone or IAD device setting you may need to set these values to see how it works exactly right for your network.  You may need to adjust the setting a bit.Example:-
    Min Delay:  120
    Max Delay: 120
    Normal Delay: 120
    You may try to change the type “adaptive” to “fixed”. Contact your IP Phone or IAD device vendor for support.

NOTE: Some helpful tools here you may like to use to help on your troubleshooting.

  • wireshark: https://www.wireshark.org/
  • ntop: http://www.ntop.org/

PROBLEM #3: I hear an echo.

Echo is the sound repeated of your own voice at a later interval.  It is usually caused by one of the following:

REASONS:

  • Acoustic feedback from your voice going out of the mouthpiece at the other callers end and entering back into the mouthpiece.
  • Other devices at the near end of the Office phone or IAD, such as a separate caller ID or a splitter that can create an impedance mismatch.
  • Electrically through poor quality phone lines.

FIX:

Adjust the volume of the phones down to low to prevent feedback.  Take out any separate caller ID’s or splitters that could be responsible for an impedance mismatch causing near end echo.

PROBLEM #4: I hear static.

Buzzing noise or static is most likely the result of analog interference into the phone lines or phone.  This electrical interference can be caused by any of the following:

REASONS:

  • An electrical voltage being added into the lines by an alarm system, separate caller ID or faulty wiring.
  • Interference created from wireless devices into a cordless phone.
  • Weak signals being generated from cordless phones.
  • Using the wrong power supply for the Office phone, IAD or ATA device.

FIX:

Ensure that cordless telephones are charged and have batteries in the handset that are not weak.  Change channels on the phone to see if one is better than another.  Make sure you are using the correct power supply for Office phone device.

PROBLEM #5: Calling Issues with Non-Manage LAN Switch and ASUS Router

Phones connected behind ASUS Router via Non-manage LAN Switch have intermittent no ringtone when dial outgoing calls and sometime need to wait 5 min before  making another phone call.

Router device from ASUS does not have a firewall which masks the contact address of SIP packets. This means that SIP traffic and RTP traffic pass through without interference and should work with myPBX / myVoice without issue.

Look to the following as most probable:

REASONS:

  • Router SIP ALG broken.
  • Router Firmware is broken causing blocking of signaling to the Office phone device, failure of router to translate NAT correctly.
  • LAN switch do not have QoS for SIP/RTP or failure to handle VoIP traffic correctly.

FIX:

  • Separate your Data and Voice by using VLAN via Manage Switch with VoIP QoS.
  • Dedicate a Manage LAN Switch that support VoIP QoS for your VoIP device.
  • Upgrade or Downgrade Router Firmware to the version that do not breaks VoIP traffic.
  • Fix NAT handling by working on Router WAN port forwarding to each IP phones IP address and port.
  • If there is a SIP ALG running, Disable SIP ALG.
    ================================
    To disable SIP ALG on your Asus router just log in to the routers GUI and and do the following…SIP ALG is located in (via the web interface):Go to Advanced Settings / WAN on left side.
    From the tabs across the top, choose NAT Pass through.
    Change SIP pass through to “Disable.” Hit apply.For phones to pick up the change immediately, reboot each of them, otherwise they will pick up the new NAT table with changes during their next registration.

    If your router does not have an option to disable SIP Passthrough then read on…

    To disable the SIP ALG manually, you enable telnet to the device via the WWW interface.
    Telnet to the device (from a command line enter “telent 192.168.1.1” or the appropriate IP address for the device.) and enter the following commands:

    nvram get nf_sip

    (It should return a “1”)
    nvram set nf_sip=0
    nvram commit

    Then reboot the router for the changes to take effect.

    For phones to pick up the change immediately, reboot each of them, otherwise they will pick up the new NAT table with changes during their next registration.
    ================================

PROBLEM #6: Calling Issues with router firewall

If your VoIP telephone is placed behind a router or a combined modem/router, you may experience problems with your myPBX/myVoice service.

Some of these problems include:

  • VoIP light goes out intermittently or constantly.
  • You cannot make outbound calls or when making outbound calls you hear a busy signal even though the person you are calling hears a ring.
  • Incoming calls go directly to voicemail without ringing your VoIP phone.
  • You hear a busy signal in the middle of the call.
  • Your ongoing calls for estimated 10 minutes and you get call cutoff (drop calls).
  • You experience quality issues such as choppy conversation or one person can’t hear the other.

REASONS:

  • These issues are often due to your router’s firewall (also known as NAT) blocking certain operations of the VoIP telephone adapter.
  • Router Firmware is broken causing blocking of signaling to the Office phone device, failure of router to translate NAT correctly
  • LAN switch do not have QoS for SIP/RTP or failure to handle VoIP traffic correctly.

FIX:

Try the following solutions to resolve the issues. Please make changes one at a time and reboot your router and VoIP device each time to see if the problem is solved.

  • Please check that you have Internet connectivity (try to view a few web sites) and check if your router/firewall settings have changed.
  • IMPORTANT Upgrade the firmware (firmware is similar to software) on the router. Most VoIP firewall isues are resolved by router firmware upgrade. For detailed instructions consult your router’s user guide or the manufacturer’s web site.
  • Some routers alter SIP packets with the default configurations which creates VoIP service problems. From your router’s web configuration page (usually under configuration / firewall / advanced settings);
    • Disable Stateful Packet Inspection (SPI) if applicable.
    • Disable SIP Application Layer Gateway (SIP ALG) if applicable.
  • Try disabling your firewall (turn it off completely) briefly. Reboot your router and VoIP device and check if you can make/receive calls. If you can do so now then your problem was with your routers firewall configuration. You firewall is not allowing calls to your SIP phone. You can continue using your router as firewall disabled otherwise consult your router vendor.
  • Enable the DMZ option on your router which will open the firewall for one specific host. To determine the IP address or host (your VoIP device) to enable DMZ for, you will need to login to your router and look at the devices connected to it. From there you can get the internal IP address or host information that it has assigned to your VoIP device. For detailed instructions consult your router’s user guide, the manufacturer’s web site.
  • Disable DMZ and try forwarding only VoIP ports on the router to your VoIP device. The following ports are needed for VoIP communications from your VoIP device to the myPBX/myVoice servers. Consult routers manual or call their technical support to forward these ports on your router/firewall:
    (*) port 5000-5500 for UDP and TCP
    (*) port 10000-20000 for UDP