Recommended firewall settings on customer's router
| # | Protokol | Transport | IP:Port(y) | Traffic direction | Note |
|---|
| 1 | SIP | UDP | PBX_hostname:5060 | Outgoing, Incoming* | Call signaling |
| 4 | RTP | UDP | PBX_hostname:5004 - 20000** (Aastra/Grandstream) | Outgoing, Incoming* | Media |
| 5 | HTTPS | TCP | PBX_hostname:8089 | Outgoing, Incoming* | Communicator App |
| 7 | STUN/TURN | TCP,UDP | 212.71.175.51,212.71.175.52:3478, 3479, 5349 | Outgoing, Incoming* | Support for NAT traversal |
| 8 | HTTPS | TCP | PBX_hostname: 443 | Outgoing, Incoming* | Web interface (UI) |
| 9 | NTP | TCP,UDP | 212.71.128.13:123, 212.71.175.14:123 | Outgoing, Incoming* | Time server |
*) Incoming communication within an already established connection, comes from PBX (service) from the specified source port.
**) Depends on end device, described example is for Aastra a Grandstream.
Communicator application (webcom.ipex.cz)
| Name | Protokol | Description | IP:ports |
|---|
| Google Firebase | https://communicator.firebaseio.com/ | Real-time database service | The service establishes a websocket connection |
| Voice traffic - data | DTLS-SRTP | Encrypted voice communication | IP_PBX:dynamic port ( Can be limited by Chrome policy ) |
| Voice traffic - signaling | WebSocket | Encrypted call signaling | IP_PBX:8089 |
| WEBCOM API | https://webcom.ipex.cz | Webcom application and API |
|
| B2B API | https://restapi.ipex.cz | B2B API |
|
| PBX API | https://ipbxapi.voipex.io (213.168.161.130, 213.168.163.87, 213.168.163.2) | PBX API for management |
|
| Statistics | https://ipbx-stats.voipex.io | Call center statistics |
|
| Meetings API | https://meeting.voipex.io | API for videoconferencing |
|
| Amazon S3 |
| User photos |
|
Web application for pbx management (name.voipex.io)
SIP connector (SIP trunk)
Definition on central PBX IPEX: 212.71.129.36
| Protocol | Transport | Ports | Codecs | Traffic direction |
|---|
| SIP | UDP | 5060 | G.711a, G.729, G.722, H.264 | IN/OUT |
| RTP | UDP | 5000-20000 | G.711a, G.729, G.722, H.264 | IN/OUT |
Requirements for calls to the quality of the Internet line
| Name | Standard | Value | Description |
|---|
| Delay | ITU-T G.114 | < 150 ms | Network packet delivery delay |
| Jitter |
| < 30 ms | Variation in the amount of packet delay when passing through networks |
| Packet loss |
| < 1 % | Number of packets lost in percent |
| Bandwidth |
| 90Kbps | Data bandwidth for one speech channel (symmetrical, G711 codec) |
Requirements for video conferencing calls - bandwidth
Theoretically, there is no minimum data rate for video calls, as the picture quality is automatically adjusted to the current capacity of the Internet connection (ie it can change during the call), however, the following applies to maintain a certain quality:
- 1: 1 video calling - 600kbps (up / down) for HQ quality and 1.2Mbps (up / down) for HD quality
- For group video calling: 600kbps / 1.2Mbps (up / down). For tile display: 1.5Mbps / 1.5Mbps (up / down).
- Screen sharing only (no video thumbnails): 50-75kbps
- To share the screen with video thumbnails: 50-150kbps