The basic IP settings are made here.
The TOS (Type of Service) byte inside the IP header is used for prioritization of packets inside the network. The device allows some packets to be marked as higher priority. The ToS values for this purpose are configured here and must be provided as hexadecimal values for 8-bit field of ToS. See related articles for converting from well known decimal DSCP values.
|TOS Priority RTP-Data||Configuration of the TOS value for media (e.g. voice) packets. The default value is 0xb8 (DSCP Expedited Forwarding according RFC 3246). Consequently, voice data receives priority forwarding.|
|TOS Priority Signaling||Configuration of the TOS value for Signaling (e.g. H.323 or SIP) packets. The default value is 0x68 (DSCP Assured Forwarding according to RFC 2597).|
RTP Port Range
First UDP RTP port / numbers of port: This entry restricts the range of ports in which UDP RTP voice data (User Datagram Protocol Real-time Transport Protocol) is received for H.323 or SIP calls. The port range 16384 to 32767 is used as standard.
Range calculation: 4 * <estimated-number-of-simultaneous-calls> but at least 128.
Note: For every single VoIP connection 4 ports are allocated. One (even) RTP port, one (odd) RTCP port, one (even) T.38 port. The fourth port is not used but lost, since the next call's RTP port must be an even port again. This information applies to all kind of devices, even such devices that do not support T.38 fax (telephones).
Restrictions: 128 ports are the smallest range and 16384 it's the max range.
UDP NAT Port Range
First UDP NAT port / numbers of port: This entry restricts the range of ports that may use UDP NAT data (Network Address Translation). There is no default for this entry. If it is left empty, no UDP NAT will be done. It is recommended to keep the UDP NAT range distinct from the RTP port range.
Note: The minimum number of ports that must be configured it's 100, lower values will not be accepted.
Here you can declare ip addresses or address ranges to be part of the local network.
This configuration influences the coder selection process on VOIP endpoints like telephones and physical Gateway interfaces.
See Reference:Configuration/Registration/Registration for further information regarding coder preferences.
Here you can declare ip addresses or address ranges to be part of the private network.
This configuration influences media negotiation in the Gateway application and the PBX application.
If the Gateway/PBX forwards a call between private and non-private network addresses (inter-network calling) the Media-Relay mode is used. This keeps private IP addresses from being handed out to endpoints in public network which cannot send to private IP addresses.
If the Gateway/PBX forwards a call between two private or two non-private addresses the Transit mode is used.
In Media-Relay mode RTP addresses are NOT passed through between the calling endpoint and the called endpoint. If the Gateway/PBX allocates local RTP ports and announces it's local RTP address to both endpoints. In result both endpoints send their RTP towards the Gateway/PBX from where it is relayed to the opposite endpoint.
In Transit mode both endpoints are allowed to exchange their RTP addresses and send RTP directly between each other. In this mode the media streaming is de-centralized which perfectly scales up to a very high call density.
Please note that Media-Relay mode is currently the only way to interop RTP-DTMF between H.323 and SIP devices.
Howto:What Ports are used for Signaling and Voice Traffic in SIP and H.323?
Howto:The IPv4 TOS field and DiffServ
Howto:Calculate Values for Type of Service (ToS) from DiffServ or DSCP Values