A fully equipped Call Centre, direct from the cloud, get started today
Unlimited call recording to help you manage compliance and performance
Bring your teams together with our collaboration features, including video calls and video conferencing.NEW
Mostly inbound calls?
Choose a low-cost monthly licence fee and simply pay as you go for the calls you make
Outbound calls to different places?
Get 2,000 outbound minutes per user/ month and one inbound number per user
Covers international calls to over 45 countries including:
In an ideal world all devices on the Internet would be able to communicate directly (roll out of IPv6 promises to make this possible with almost unlimited addressing space). However as of today most of us still use good old IPv4 which means all our private networks are behind NAT (Network Address Translation) device.
To help network administrators configure their NAT equipment to allow SIP phones (or softphones) to communicate with VoIPstudio network (SIP proxy servers and RTP media gateways), below you can find call flow diagrams showing network addresses and ports involved. In diagrams below green lines indicate SIP (Session Initiation Protocol) used for signalling (call set up and tear down), blue lines indicate RTP (Real-time Transport Protocol) which transmits audio streams during a call.
Once SIP phone is powered on it will attempt to register it’s location with VoIPstudio network. This is to allow routing of incoming calls to specific endpoints on the network. It also creates NAT binding which is than kept open thanks to periodic “keepalive” packets which are sent between SIP phone and VoIPstudio servers.
Figure 1. SIP NAT Traversal – REGISTER
Note: there can be more than one SIP phone on Private LAN, as NAT Router will create a unique random WAN_IP:port binding for each device as shown (2) in Figure 1 above.
Normally NAT device would close NAT binding created in step (2) Figure 1 above after a short period of inactivity (usually 60 – 900 seconds depending on the device). This would make impossible for VoIPstudio servers to reach phone and alert it on incoming calls. To keep NAT binding open, we use SIP Keepalive technique, which sends SIP OPTIONS packet (which has no function other than make the SIP phone reply to it with SIP OK) every 30 seconds.
Figure 2. SIP NAT Traversal – Keepalive
When making an outbound call SIP phone will send SIP INVITE packet to VoIPstudio server which is challenged for user credentials and re-sent. After successful authentication VoIPstudio server responds with SIP OK packet that includes information about RTP (media) server public IP and UDP port number where phone should send it’s audio stream.
Figure 3. SIP NAT Traversal – Outbound Call
When routing an inbound call VoIPstudio SIP server uses terminal location (public IP address and port number) information stored during Registration process shown in Figure 1 above.
Figure 4. SIP NAT Traversal – Inbound Call
Note: time between steps 3 and 6 described above is between 20-50ms, therefore there is no noticeable silence at the beginning of the call.
Most modern NAT Routers and SIP phones will work as presented in above diagrams “out of the box”. However if you notice problems with inbound calls, one way audio or other unusual behaviour, please make sure your equipment is configured as below:
Start a free 30 day trial now, no credit card details are needed!
Thousands of businesses across the world trust VoIPstudio for all of their most vital business communications. Why not be the next?
Thousands of businesses across the world trust VoIPstudio for all of their most vital business communications. Why not be the next?
Start a free 30 day trial now, no credit card details are needed!