Lately I have had the opportunity to try a SIP video phone unique: it is produced by the SVP 5000 SBNTech.
The key features of this product are:
- Videophone Calls over the Internet
- LCD Touch Screen (480 × 10.2 pollici/800)
- Conform to IETF, ITU-T
- User-friendly interface
- Video calls @ 30 FPS
- Digital Photo Frame as a Screen Saver
- Web Browser
- SD Card Slot & USB 2.0
- WiFi (802.11b/g/n)
- Bluetooth (option)
- An FXS port
- Integration HSDPA
- Video / Audio Input / Output
- Camera / LCD Power Save Mode
- Support for audio codecs g711/g729a
- Support for video codecs h263 and h264 (QCIF / CIF / QVGA / VGA)
The tests were conducted using both Asterisk Asterisk 1.6.x to 1.4.x on CentOS Ubuntu Server in both cases installed from source. The configuration used for the tests was the following:
videosupport=yes; abilita il supporto video in asterisk
In principle, I suggest you avoid, as far as possible a setup where you have to worry about Asterisk retransmit the audio / video streams to the two end-points. This is the reason why, in the suggested configuration, the parameter "canreinvite" is set to "yes".
The parameter "canreinvite = yes" makes it possible to Asterisk, once the call has been accepted, to send a new INVITE to the two endpoints with the information necessary for establishment of a audio / video communication directly between them.
Please note that this "reinvite" does not happen, despite the "canreinvite = yes" if:
1) one of the two clients is configured with "canreinvite = no";
2) if the clients are using different codecs.
3) If the command Dial () contains the parameters for recording sound (wo W) / call transfer (to T) / forced hangup (I H).
In all cases, the bandwidth required for a video call and of at least 200Kbps, always that the canreinvite is active. In other cases need more bandwidth. Attention to the fact that as soon as an active video call, the phones use a value of bandwidth as low as possible. To get higher quality video you must manually increment the bandwidth parameter dall'apposito configuration panel in the call.
In my testing I used: H264 / G729 / VGA. It 's always possible to scale the parameter VGA to QVGA or QCIF if you want to further decrease (although not by much) the required bandwidth.
The tests carried out also seems that only when the work canreinvite a video call between the phone and a desk phone is not activated in the first video. In other cases, it goes hand-chosen not to activate the video when you make or accept a video call from the phone.
Very intuitive and simple user interface: All functions are easily accessible thanks to the convenient touch screen.
The product was kindly provided by www.4geek.it
Manufacturer's page: SBNTech
Tags: Call Center Systems | VoIP PBX | Asterisk Consultant Naples | Switchboard Phone | Voip | Asterisk CTI | PBX | IP Phones | Networking | Linux
Development of IVR, call center systems, Voip PBX.