Posts tagged: Siemens Gigaset SIPPHONE Gizmo5 Settings

Siemens Gigaset SIPPHONE Gizmo5 Settings

By , December 23, 2010 7:23 PM

I have searched numerous forums and cannot find any decente
sipphone settings for the Siemens Gigaset Phone for Sipphone / Gizmo5

The setup:

Personal Provider Data
* Authentication Name:  1747xxxxx what ever your gizmo5 number is
* Authentication password: password what ever yours is
* Username:     1747xxxxxx what ever your gizmo5 number is
* Display name:  1747xxxxxx what ever your gizmo5 number is

General Provider Data
* Domain:   proxy01.sipphone.com
* Proxy server address: proxy01.sipphone.com
* Proxy server port:     5060
* Registration Server: proxy01.sipphone.com
* Registrar server port: 5060
* Registration refresh time: 180

Network
* STUN enabled: YES
* STUN server: stun01.sipphone.com
* STUN port: 3478
* STUN refresh time: 240
* NAT refresh time: 20
* Outbound proxy mode: Never (can anyone confirm?)
* Outbound proxy: EMPTY
* Outbound proxy port: 5060

If you are having trouble with missing one part of the audio on a call ie cant hear the other person speaking but can hear and speak
fine otherwise then you should enable STUN.     (can anyone confirm?)
Gizmo5 does not provide an outbound proxy, so you should leave that field blank.

The screen shot:

siemens gigaset sipphone gizmo5 setup

The codecs:

Codecs:
GSM — fixed bit rate, not loss tolerant, narrow band (8khz sampling rate).
iSAC — variable bit rate, loss tolerant, narrow and wideband (8 to 16khz). Varies based on Bandwidth, packet loss, delay
iLBC — variable bit rate, loss tolerant, narrow
PCMA — fixed bit rate (8kHz sampling rate)
PCMU — fixed bit rate (8kHz sampling rate, high band width)
IPCMWB — 16 kHz sampling rate
EG711 (enhanced g711) — fixed bit rate, loss tolerant, narrowband
iPCM — fixed bit rate, loss tolerant, wide band.

I would recommend that you enable
ulaw u711
alaw a711
GSM
iLBC
g729

and disable
g722 (wideband)

I believe but have no confirmation that it does g729 in passthu mode only. (can anyone confirm?)

DTMF

In-band
Incoming stream delivers DTMF signals in-audio independently of codecs

Out-of-band
Incoming stream delivers DTMF signals out-of-audio using either SIP-INFO or RFC-2833 mechanism,
independently of codecs – in this case the DTMF signals are sent separately from the actual audio stream.

SIP-INFO is not recommended for DTMF delivery, since it cannot deliver strokes synchronously with the audio stream, introducing timing artifacts (mainly because it’s delivered using SIP, which is not a real-time mechanism for delivering media). It is very common for public services to NOT support SIP INFO, and it seems unlikely that such services will improve support for this delivery mechanism.

We want to use rfc2833

Thomas Challenger Thomas Challenger