- If your SIP registration failed, please check the following:
- Double confirm that your internet is working well. To check the internet you can:
- Launch web browsers like Chrome and address to google.com or bing.com.
- Ping google.com or ping sip1.b3networks.com (or the SIP domain we provide).
- Login to your account > click the SIP app > check to confirm if you are using the correct SIP credentials. You have to check the following information and make sure your configuration must be matched:
- SIP username
- SIP Password
- SIP Server
3. Double-check your NAT and firewall settings and ensure that they follow the settings on the SIP subscription page. If you make any settings changes at this step, remember to restart the device to ensure that the new settings are deployed.
4. Ensure you remove all expired SIP trunks, In the case you try to register expired SIP trunks, we treat that as a hacking case then block all traffic that comes from your device via public IP.
- A jitter buffer temporarily stores arriving packets in order to minimize delay variations. If packets arrive too late then they are discarded. If a packet was dropped (or simply does not arrive in time) then the receiving device has somehow to “fill in” the gap using a process known as Packet Loss Concealment or PLC.
- Packet loss needs to be less than 1% if it is not to have too great an impact on call audio quality. Greater than 3% would certainly be noticeable as degradation of quality (The default G.729 codec requires packet loss far less than 1 percent to avoid audible errors. Ideally, there should be no packet loss for VoIP)
Please run the MTR, which is a powerful network tool enabling administrators to diagnose and isolate networking errors and providing helpful reports of network status to upstream providers.
- MRT-rw-c50 sip1.b3networks.com (or the SIP domain in your SIP subscription page)
III. Bandwidth Problem
- G.729: 24Kbps upstream; 24 Kbps downstream
- GSM: 29 Kbps upstream; 29 Kbps downstream
- G.711: 80 Kbps upstream; 80 Kbps downstream
Is your internet connection asymmetric?
IV. Hardware Issue
- Router: Many small businesses use their internet connection for both voice and data. This is perfectly fine as long as your router has the ability to prioritize VoIP traffic. Without a router that is configured for 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 prevents this from happening by giving priority to voice traffic on your network.
- SIP Device (PBX or IP Phone): Sometimes the SIP hardware goes to crash or performance overload. In order to confirm the issue on hardware, just reboot or use the soft-phone to see how voice quality goes.
- The bandwidth required per SIP calls for more details. If your router provides network statistics, you can easily investigate if your internet capacity is utilized at or near the maximum provided by your Internet Service Provider (ISP)
- If you can hear the other party clearly and the other party cannot hear you, it is most likely that your internet upload bandwidth is not sufficient. Double-check with your ISP on your upload bandwidth as it may be much smaller than the download bandwidth speed.
- Connected Devices in the SIP subscription page to double confirm that only ONE device is connected. A SIP account can ONLY be used with one device. If more than one device is connected, you may experience no audio or one-way audio.
- Check NAT setting configured properly on SIP devices and Routers. The settings can be found within the SIP App configuration page. If you make any settings changes at this step, please remember to restart the IP-Phone and IP-PBX to ensure that the new settings are deployed.
- Confirm that SIP ALG is disabled. Many of today’s lower-end commercial routers implement SIP ALG coming with this feature enabled by default. While ALG could help in solving NAT related problems, the fact is that many routers’ ALG implementations are wrong and break SIP. We will not support SIP ALG.
- Please consider setting for TURN & STUN server
- Confirm that you are using one of the supported codecs: G.711 (alaw, ulaw), G.729, GSM. Consider testing with each other codecs (with ptime = 20ms)
- Confirm your Network is in good condition (Sufficient Bandwidth, Latency, Packet Loss …)
- If your account is a prepaid account, please confirm that you have a positive balance in your account.
- Confirm that you have sent the invite to the correct IP address of a SIP server.
- Confirm that you are specifying the called number in E.164 format including the country code. For example, +6566021000 for calls to Singapore regardless of where you are originating the call.
- The called number is in service
- Did you configure any security settings?
- IP White-list
- Country White-list
- Make sure your Firewall doesn’t block traffic from us
- Your Network in is good conditional
- Confirm that you are using one of the supported codecs: G.711 (a-law, u-law), G.729, GSM
- Double confirm your device register to the SIP trunk successfully
- Successful register message show on your device or
- Login to your account, check Connected Devices to double confirm that the correct device is connected.
- A SIP account can ONLY be used with one device. If more than one device is connected, you will NOT be able to receive incoming calls at the correct device. The call will be shuffling.
- Check to confirm that you are using one of the supported codecs: G.711, G.729, GSM
- Check your Firewall Settings
- Almost all cases in which lines are unable to receive incoming calls are due to network latency fluctuations. UDP transport cannot handle very well in this case. If your SIP devices are still using UDP transport. Please switch to TCP transport.
- If you have completed the above and it still does not work, please collect the MTR Report and Network Capture (or SIP log) of a failed incoming call. Send both of them to our support team for further assistance.
COLLECT MTR REPORT
1. Using MTR on Unix-based Systems (Linux/GNU)
For example, to test the route and connection quality of traffic to the destination sip1.b3networks.com (or the SIP domain in your SIP subscription page), with useful flag -rw -c50, input following command
root@ip-10-7-1-231 ec2-user]# mtr -rw -c50 sip1.b3networks.com (or the SIP domain in your SIP subscription page)
2. Using MTR on Windows SystemsFor Windows, we use the WinMTR.
3. Using MTR on MAC OS XTo gather the MTR report on MAC OS X, you need to install MTR package at Rudix. Then run the following command in Terminal:sudo /usr/local/sbin/mtr -rw -c50 sip1.b3networks.com (or the SIP domain in your SIP subscription page)NOTES:
- The destination host should be the SIP domain your SIP account is based on.
- To understand more about MTR, please refer to Linode.
COLLECT NETWORK TRAFFIC CAPTURETo capture the network traffic, we using Wireshark/TShark tools. This is a free and open-source packet analyzer. It is used for network troubleshooting, analysis.
1. Using TShark on Linux/GNU SystemsSyntax of command, input tshark -h for Usage instruction
- tshark [options]For example, capture the network traffic on NIC0 and save the capture file with the name “capture-output.pcap” enter the following command:
- tshark -i eth0 -w capture-output.pcapPress Ctrl + C to stop capture network traffic.
2. Using Wireshark on Windows (download). Note that select the correct interface to capture network traffic.
- The originating telco may not send the correct format or the original From number.
- A caller ID may be modified along the way by any intermediary telcos
- The caller ID may be lost or replaced when crossing international borders
- You will not receive Singapore mobile CLI as the CLI will be replaced by other CLI (ie a random local number) according to Hong Kong CLI regulatory. The CLI will be replaced by the 1st HK terminated operator.
- A Singapore Number dials to a HK number > (1st) HK operator replaces Singapore CLI according to HK CLI regulatory > CLI by regulatory > Customer side.
- NOTE: According to HK CLI regulatory, CLI for HK is mobile only. There is no CLI allowed on the HK Landline.