The following steps can be used as a guideline to help resolve issues with Phone.com services:
- Identify the issue.
- Establish a theory of the probable cause.
- Correct the problem.
- Test to make sure it is fixed.
1) Validate that a problem exists:
- Gather detailed information on the reported issue.
- Replicate or reproduce the issue, if possible.
- Check devices’ status, registration, and provisioning time stamp.
- Is this a known issue or an emerging problem with the Phone.com services? You can check here: https://status.phone.com/.
2) Establish the cause of the problem
- Question the Obvious – Does it have power? Is it connected properly?
- Any network changes?
- ISP (Internet Service Provider)/ Internet access operational?
- User Error?
- Did it ever work before?
- Check call handling rules
3) Correct the cause of the problem.
- The phone was plugged back in
- It was reconnected to the router
- New network devices re-configured
- ISP was down
- Reboot devices/network
- Fixed call handling rules.
4) Test that the issue was in fact, resolved
- Place a test call or verify that the issue is fixed.
- If possible, try to find a way to prevent the issue from repeating.
If, after troubleshooting and attempting to correct the problem, the problem is not resolved, it may need to be escalated to our preferred support team.
Below are some common situations that you may encounter along with some ideas of what to do:
If a device on the network shows offline:
- Find the IP Address of the phone.
- If the phone does NOT have an IP address, check connectivity (Cables, Connections) • If still no IP, Plug the phone directly into the router (if not already)
- If still no IP, Reboot the router.
- If the phone still does not have an IP address, the customer will have to reach out to the ISP, or Network Administrator for help.
Calls Quality Issues
- Is there audio quality issues only in certain areas? I.e. does the call quality get bad if they are near bad wi-fi or cellular data areas?
- If the end-user has a bad cellular data signal and the audio quality is poor due to it, you can have your customer press the options icon on the bottom right, then have them press Switch to Carrier. This will allow them to use their voice signal instead of their data signal.
- If the end-user is moved to Gateway mode (Basically, are we using their voice network over their data/wi-fi network), does the behavior still exist for calling?
Were you able to verify if the permissions that are needed were accepted?
- iOS will need to enable Push Notifications and Microphone
- Android will need to enable the Ability to Make/Receive Calls and the ability to Record Audio/Microphone
A network is a group of computers and other devices, such as IP phones, linked together. In this section, we’ll cover the types of devices that make up a network.
It is necessary to have a fast enough and reliable internet connection in order to get the best quality of service from VoIP. Before implementing VoIP, we recommend making sure your internet can handle the increased use of bandwidth.
Bandwidth is the amount of data that can be transmitted in a fixed amount of time. This is expressed as an upload and download speed in Mbps (megabits per second), and lets you know how much data you have to work with.
You can check your speed here: http://www.speedtest.net/
The table below shows the minimum bandwidth required to make calls, as well as the recommended speeds for optimal performance.
|Concurrent Calls||Minimum Required Bandwidth||Recommended Speed|
|1||100 Kbps Up and Down||3 Mbps Up and Down|
|3||300 Kbps Up and Down||3 Mbps Up and Down|
|5||500 Kbps Up and Down||500 Kbps Up and Down|
|10||1 MBps Up and Down||5-10 Mbps Up and Down|
Typically provided by your ISP and connects you to the internet. Modems with built-in routers should be avoided. (See below)
Very important for VoIP to work properly. It routes all the traffic within a local network (LAN) to external networks. It also provides security, allows you to share your network, and creates a private network for all connected devices.
It’s not uncommon for ISPs to provide a modem with a built-in router, also known as a modem/router or “gateway”. These particular devices should be avoided as they are not always compatible with VoIP traffic and/or may require changes.
If there is a gateway, it is recommended to purchase a compatible standalone router. Then the gateway should be bridged to this router. This is done by disabling the routing portion of the gateway. The ISP should do this.
Now that we understand what a network is and the devices that make one up, let’s take a look at how a phone call works.
VoIP applications use two protocols, SIP (Session Initiation Protocol) and RTP (Real-time Transport Protocol). SIP is used for establishing and terminating a session (phone call). After the call is established, RTP takes over and is used for sending the voice data (packets) between phones. A packet is just a unit of data that carries information across networks.
Think of SIP as the stage manager. SIP prepares the stage for RTP by setting up its connection. Once RTP is finished with its stage (call), then SIP comes back to the stage to clean up after it.
Since these are different from protocols used in email (SMTP), streaming music and videos (RTSP), and browsing the internet (HTTP), oftentimes, routers and firewalls are not configured by default to work with VoIP, and the traffic is blocked, or the voice packets will not be routed properly to our server and back.
SIP uses ports 5060 and 6060, and RTP uses ports 10000-49999. If these are blocked, the packets won’t be able to leave the network and ultimately get to where they need to be (our server). Think of a roadblock and the cars (packets) not being able to pass.
When the voice of either or both people on the phone cuts out during the conversation, sounds distorted or muffled, or there is a significant delay between when one person speaks, and the other person hears the message. Most often associated with a problem with your internet connection or network setup…
- Lack of sufficient upload bandwidth
- High latency
- Competition from computers on the network uploading large amounts of data • Incorrect settings on your network devices
Sometimes the phones themselves can cause sound issues. If one phone is having problems, perhaps try swapping out the piece of phone equipment (handset/headset). Interference from large and small electronic appliances can also cause sound quality problems. Crossed or tangled cables can cause feedback. For example, if your headset cable is crossed or tangled with your iPhone charger.
Constant sounds like buzzing, low hum, crackles, pops, even when no one is speaking.
- Common causes of static are electrical interference from other devices. Move VoIP devices at least four feet away from other electronic devices.
- Static on only one phone could be the result of bad headset or handset.
- Check cables and connections. A defective telephone or ethernet cable or a bad port on your router can cause static.
- Regularly rebooting the modem and routers in the network is recommended to prevent issues such as static.
Network equipment’s firewall settings can be the cause of sound issues if not set to work with VoIP services correctly. Not all modems, routers, and firewalls are VoIP compatible. Using incompatible devices, settings, or configurations can result in call quality or reliability issues.
Double NAT (Double Routing)
A “double NAT” occurs when there are multiple routers on the same network doing network address translation. This is known to cause problems with VoIP applications. Ideally, only one device is needed to perform routing functions. This scenario is most common when a user has a modem/router + standalone router. It is best to eliminate or bridge extra routers or modem/router combos on the network. It is recommended for the user to contact their ISP to bridge the modem/router device. If it is not possible to bridge the device, the user will need to exchange the modem/router for a standalone modem.
Phone not obtaining an IP address
If the phone is not obtaining an IP address, then it cannot connect to the network. Try rebooting the router first otherwise, this could be caused by:
- Ethernet cable not being plugged into the correct port on the phone (Ex. SW vs PC on Cisco) • Phone not connected to the network properly
- DHCP not enabled
- Defective ethernet cable
If the customer is reporting the following symptoms, refer to the setting guidelines below for configuring their router’s firewall to properly handle our traffic.
One-Way or Two-Way Audio
Dropped sound is when either or both people on the phone cannot hear the other person. Sound is lost during an active call and does not resume.
- Dropped audio on multiple phones is usually due to incompatible network equipment or the firewall settings of the router.
- Dropped audio on a single phone can be the result of faulty telephone equipment or the volume settings on the handset, headset, or speakerphone.
There is no audio at all from the start of the conversation. Call rings and is answered, but there is no audio.
- This can be the result of faulty telephone equipment or the volume settings on the handset, headset, or speakerphone.
- It can also be caused by the phone’s connection to the network, either a bad connection or a defective ethernet cable.
It is recommended to have the customer’s IT administrator apply the following changes listed below to ensure it is done properly.
For smaller offices with off-the-shelf routers, these firewalls may need to be modified or turned off. On enterprise-level equipment, certain rules may need to be added. The IT administrator should make these adjustments.
The following settings listed below can typically be found in the router’s web interface and can be easily checked on/off.
SIP Application Layer Gateway is a setting that oftentimes prevents our traffic from flowing properly. If enabled, SIP ALG can cause various issues such as loss of connection with our servers, calls disconnecting, loss of one-way or two-way audio, phones continuing to ring after being answered, and phones ringing randomly or out of sequence. Many routers will have the option to disable this feature, usually under the Firewall section (check the specific router’s user manual). If the option to disable SIP ALG is not available, we recommend purchasing a router that doesn’t have this feature or at least allows for it to be turned off.
To ensure that traffic is not being blocked, the following ports used for VoIP communication to our servers should be opened:
- SIP Ports 5060, 6060
- RTP Ports 10000 – 49999
Other Router/Firewall Settings
- Make sure our IP is set as a trusted source: 184.108.40.206/23
- Increase UDP timeout settings up to 120
- Disable SIP Transformations
- Enable consistent NAT
SPI (Stateful Packet Inspection) / DoS Protection
SPI/DoS allows the router to approve or deny any information packets for security reasons. Oftentimes it will incorrectly identify our VoIP traffic as a security risk. DoS protection keeps track of how many connections are made to an individual web address and begins blocking access once a limit is reached. Our phones connect to the same site. The more phones in your network, the more likely SPI/DoS will begin to block connections. To prevent this, some routers will allow you to either disable SPI/DoS protection to allow more connections.