Phone web page
- If your Grandstream GXV3240 shows the video in the resolution 352x288 only, although the resolution 640x480 was configured, then configure in 2N IP Intercom menu Services / Phone / Video / Advanced SDP Settings, parameters H.264 Payload Type (1) to 99, H.264 Payload Type (2) to 0. Then your phone will show video in the resolution of 640x480.
2N IP Intercom
Call Completed Elsewhere
|Secure RTP (SRTP)|
Settings with PBX
This guide describes basic steps for configuration of peer to peer communication between 2N IP Intercom and IP phone Grandstream GXV3240. There is also a possibility to register both these devices to the IP PBX (SIP proxy server) and use internal dialling plan for calling between each other but it is not described in this FAQ.
Grandstream GXV3240 IP video phone is placed in the same LAN (local network) as intercom. IP address of GXV3240 is 192.168.50.196 and IP address of 2N IP Intercom is 192.168.50.199. These IP addresses and names are used only such as example – please change it according to your names and network plan.
How to set 2N IP Intercom?
Settings of 2N IP Intercom is very easy. First of all create a new user and assign him a phone number – we will make SIP direct call (peer-to-peer call) to GXV3240 in our scenario. Therefore the number is set in the format: sip:IP_address (or sip:x@IP_address). See the picture below for more information.
Services->Phone->Video and enable parameter Polycom compatibily mode.
How to set Grandstream GXV3240 IP phone?
The easiest way how to set up this IP phone is via web interface but you can also use touch screen of this phone and set all necessary parameters this way. Complete setting via web interface is described below. All settings are done after factory reset which is available in the "Maintenance/Upgrade" section.
SRTP on GXV3240
Go to Account settings and to Codec Settings there set SRTP Mode to "Enabled and forced" and SRTP key length to AES 128&256 bit.
Go to SIP settings in Services and in Advanced settings enable recieving and transmitting of SRTP calls.
- Verified with video
- Verified with audio only
- Work with limitation