Polycom SIP 3.1 manual Real-Time Transport Protocol Ports, Central, boot server, sip.cfg, Local

Models: SIP 3.1

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Configuring Your System

 

 

 

 

Configuration changes can performed centrally at the boot server or locally:

 

 

 

Central

Configuration file:

Specify the Application browser home page, a proxy to use, and size

(boot server)

sip.cfg

limits.

 

 

For more information, refer to Microbrowser <mb/> on page A-95.

 

 

Specify the telephone notification and state polling events to be

 

 

recorded and location of the push server.

 

 

For more information, refer to Applications <apps/> on page A-98.

 

 

 

Local

Web Server

Specify the Applications browser home page and proxy to use.

 

(if enabled)

Navigate to http://<phoneIPAddress>/coreConf.htm#mb

 

 

Changes are saved to local flash and backed up to <Ethernet

 

 

address>-phone.cfgon the boot server. Changes will permanently

 

 

override global settings unless deleted through the Reset Local

 

 

Config menu selection and the <Ethernet address>-phone.cfgis

 

 

removed from the boot server.

 

 

 

Real-Time Transport Protocol Ports

The phone is compatible with RFC 1889 - RTP: A Transport Protocol for Real-Time Applications - and the updated RFCs 3550 and 3551. Consistent with RFC 1889, the phone treats all RTP streams as bi-directional from a control perspective and expects that both RTP end points will negotiate the respective destination IP addresses and ports. This allows real-time transport control protocol (RTCP) to operate correctly even with RTP media flowing in only a single direction, or not at all. It also allows greater security: packets from unauthorized sources can be rejected.

The phone can filter incoming RTP packets arriving on a particular port by IP address. Packets arriving from a non-negotiated IP address can be discarded.

The phone can also enforce symmetric port operation for RTP packets: packets arriving with the source port set to other than the negotiated remote sink port can be rejected.

The phone can also jam the destination transport port to a specified value regardless of the negotiated port. This can be useful for punching through firewalls. When this is enabled, all RTP traffic will be sent to the specified port and will be expected to arrive on that port as well. Incoming packets are sorted by the source IP address and port, allowing multiple RTP streams to be multiplexed.

The RTP port range used by the phone can be specified. Since conferencing and multiple RTP streams are supported, several ports can be used concurrently. Consistent with RFC 1889, the next higher odd port is used to send and receive RTCP.

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Page 87
Image 87
Polycom SIP 3.1 manual Real-Time Transport Protocol Ports, Central, boot server, sip.cfg, Local