- Register works in both ways
- STUN works as client
- Authentication to remote endpoints only
- Early audio (183) works in both directions
- Caller ID works in both directions
Note: The implementation is only a small subset of many SIP features.
An experimental feature to send and receive an identification over
voice channel.
If a party answers, the ID is transmitted some seconds afterwards.
The calling party listens 30 seconds after receiving an answer message
for the ID.
Add to your extension's settings file:
dov_ident <id string without white spaces>
dov_log /path/to/log/file
dov_type pwm|pcm
dov_level 0|level
'pwm' survives analog transcoding.
'pcm' is fast and will almost not be recognised.
'level' can be used to alter default signal amplitude (100..30000).
In order to provide internal tones, a clock is used to generate
chunks of 160 samples. If no tones are provided and if audio is
bridged, it is forwarded as usual.
In order to provide early audio on SIP trunk, "tones yes" must be set
at interface.conf.
In order to receive early audio from SIP trunk, "earlyb yes" must be
set at interface.conf.
Because EFR/AMR/HR codecs use dynamic RTP payload types, it is essential
to forward the actual media types between endpoints too. These media
types are used for negotiation of codecs. A dynamic payload type is
used as given by remote peer. Locally generated payload types are used
when offering codecs to remote peer.
Since LCR does not put hands on any RTP frame when directly bridged between
OpenBSC and SIP, it will now allow all speech codecs that are commonly supported
by MS and remote SIP endpoint.
It must be noted that OpenBSC must support forwarding the codec types that
MS and remote SIP endpoints support.
Currently LCR negotiates the following codecs for GSM:
- Full Rate
- EFR
- AMR
- Half Rate
Call received on an interface can directly be forwarded to a given
destination interface, instead of routing the call through PBX
application. This way calls can be forwarded without going through
route.conf.
Currently only SIP and GSM destinations are supported. Also there
are no tones generated, if one side provides no tones, but the
other wants to receive them.
The keyword "bridge <output interface>" in interface.conf is used.
Without that keyword, incomming calls are handled as usual.
The dependency on mISDN (loopback interface) is completely removed
from GSM and SIP interfaces.
The built in bridge of LCR now forwards audio data between these
interface instances or between these instances and other instances.
Additionally both GSM BS and SIP support direct forwarding of RTP
traffic between other SIP endpoint and OpenBSC, so no traffic is
forwarded by the LCR itself. This is done by forwarding RTP peer
informations between these interface instances.
This is required to bridge traffic beween non-mISDN handlers,
such as GSM, SIP and voice box. Also it bridges traffic between
mISDN handlers and non-mISDN handlers. It is the fundamental step
to get rid of mISDN (loop interface) for non-mISDN handlers.
This is required to bridge audio e.g. between SIP and GSM without
using mISDN. There will be no limitations on 'b-channels' anymore.
Still GSM and SIP requires mISDN, but this will be changed later.
With that bridge I cleaned up some code and also removed the
MESSAGE_DATA, which is not required anymore.
This support is just a simple peer-to-peer support for basic calls.
Currently it requires mISDN_l1loop interface, as every non-ISDN
interface does. Later it will be possible to use it without.
Andreas Eversberg