Sipdroid brief evaluation

Source: Internet
Author: User
Tags dtmf tones

1. audio format:

G722 HD voice (64 kbit)-only over WLAN and 3G (requires paid account)

PCMA (64 kbit)-only over WLAN and 3G

PCMU (64 kbit)-only over WLAN and 3G

Speex (11 kbit)-always try

GSM (13 kbit)-always try

Bv16 (16 kbit)-always try

From the source code, g711 exists, but g729.

2. Gain: headset, microphone, headset

3. video quality: High 352x288 @ 360 kbit

Low 176x144 @ 192 kbit

Sipdroid currently supports basic call functions. Special PBX functions need to be provided by server configuration.

The following features are provided by the free account on

* Change number format (e.g. Convert the + codes)

* Music on hold

* Support of several modes for DTMF tones

* Support for NAT (Network Address Translation)

* Simultaneous outbound calling

* Screening anonymous callers

* Time-based routing for incoming cballs

* Attended call transfer

* Conferences

* Video partition tion (video transmission is supported by sipdroid natively)

* Trigger callback or callthru (if no suitable data network available)

* Callto Skype users


Wireless WLAN, 3G, edge (2.75g) are required ).

Elastix (asterisk) allows video phones of ekia/X-lite and sipdroid to communicate with each other, but there are also problems, for example, you cannot receive a video call from another client (because the SDP will omit video codec when the sipdroid calls an answer video)

The dependency on the server is large, and many unresolved issues show this. although someone suggested that the non-standard RTSP sent by sipdroid can be configured on the server, it is also limited to sipdroid.

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