7 Mar
2012
7 Mar
'12
7:32 p.m.
Nik,
You could check whether this is PCMA or GSM-FR with RTP stream
bitrate. Capture the stream with Wireshark and look at RTP payload
size or use a voice call information dialog (Telephony->VoIP Calls).
PCMA bitrate is 64kbit (often 160bytes per 20ms), while bitrate GSM-FR
is 13.2kbit (usually 33bytes per 20ms).
On Wed, Mar 7, 2012 at 21:36, Nik Pakar <nikpakar(a)gmail.com> wrote:
It seems BSC is sending payload type GSM to LCR, but
LCR send payload type
PCMA on the sip channel.
07.03.12 22:45:03.907 CH(69): New call ref LCR<->BSC callref new=0x80000029
07.03.12 22:45:03.907 CH(69): Codec negotiation LCR<->BSC bearer
capa='given by MS' speech version='AMR given' ignored='Not suitable
for
LCR' version='5 given' ignored='Not supported by LCR'
version='EFR given'
ignored='Not suitable for LCR' version='Full Rate given'
version='Half Rate
given' ignored='Not suitable for LCR'
07.03.12 22:45:03.908 CH(69): MNCC_SETUP_IND LCR<->BSC calling
number=07777201 imsi=413011492012312 dialing number=4290080001
07.03.12 22:45:03.908 CH(69): MNCC_LCHAN_MODIFY LCR<->BSC speech
version='Full Rate given' mode 0x01
07.03.12 22:45:03.908 CH(69): MNCC_CALL_PROC_REQ LCR<->BSC progress
coding=3 location=1 descr=8
07.03.12 22:45:03.908 CH(69): unknown LCR<->BSC
07.03.12 22:45:03.908 CH(70): NEW handle handle new=0x8d65cc0
07.03.12 22:45:03.908 CH(70): INVITE from uri=sip:07777201@192.168.1.30 to
uri=sip:4290080001@192.168.1.25:4757 rtp ip=103.10.172.30 port=30026,30027
payload=PCMA:8
07.03.12 22:45:03.930 CH(70): RESPOND respond value=183
07.03.12 22:45:03.930 CH(70): Payload received rtp payload=PCMA:8
payload=telephone-event:101
07.03.12 22:45:13.117 CH(69): MNCC_DISC_IND LCR<->BSC cause coding=3
location=0 value=16
07.03.12 22:45:13.148 CH(69): MNCC_REL_REQ LCR<->BSC
07.03.12 22:45:13.148 CH(70): CANCEL cause value=16
07.03.12 22:45:13.169 CH(70): RESPOND respond value=487
On Wed, Mar 7, 2012 at 3:43 PM, Nik Pakar <nikpakar(a)gmail.com> wrote:
--
Regards,
Alexander Chemeris.
Hi Andreas, is the LCR actually transcoding gsm-fr to alaw towards sip
side ?
Rgds
Nik
On Wed, Mar 7, 2012 at 3:17 PM, Nik Pakar <nikpakar(a)gmail.com> wrote:
Hi Andreas,
Call signalling all works fine right through out from NITB to LCR and out
on SIP. how ever im getting some strange media behaviour.
My setup is,
[MS]---[nano.BTS]---[NITB/LCR]----[SIP Softswitch]
LCR is setup to bridge two interfaces, so what ever comes from gsm goes
to sip and what ever comes from sip goes to gsm.
Now a test call from a mobile to mobile, should go all the way to the
softswitch and come back.
All works fine in terms of signalling.
But in media, LCR seems sending initial SDP to the softswitch as PCMA:8
not gsm FR.
So softswitch expect the media as PCMA and not transcoding.
Same if the call goes out from softswitch, still no medial as it think
incoming media from LCR is on PCMA.
Any idea about this ?
This is the LCR trace - http://pastebin.com/5PNKYc5m
This is the sip trace from softswitch - http://pastebin.com/cVtx1mFB
Rgds
Nik
On Tue, Mar 6, 2012 at 3:38 PM, Nik Pakar <nikpakar(a)gmail.com> wrote:
Hi Andreas, got it working both ways now. Many thanks for nice work.
I will now test it further with transcoding.
And start on the documentation.
Rgds
Nik
On Tue, Mar 6, 2012 at 12:51 PM, Nik Pakar <nikpakar(a)gmail.com> wrote:
>
> Hi Andreas, i now have calls coming from external sip->LCR->gsm
>
> But still cant figure out the dial plan to send a call out on sip,
> gsm->LCR->sip
>
> I tried,
>
> dialing=072333444 : extern interfaces=sip prefix=072333444
>
> But on LCR trace it shows as below. I think im missing some thing small
> here. Appreciate if you can give a little hint.
>
> Thanks
> nik
>
> 06.03.12 12:40:41.286 EP(1): ACTION (match) action goto line 11
> 06.03.12 12:40:41.286 EP(1): ACTION goto/menu (change to) ruleset
> extern dialing 072333444
> 06.03.12 12:40:41.286 EP(1): ACTION (match) action extern line 28
> 06.03.12 12:40:41.286 EP(1): ACTION extern (calling) number 072333444
> interfaces sip
> 06.03.12 12:40:41.287 EP(1): SETUP ACKNOWLEDGE to CH(1)
> 06.03.12 12:40:41.287 EP(2): CHANNEL SELECTION (found given interface)
> interface sip
> 06.03.12 12:40:41.287 EP(2): INTERFACE (has no function) interface�@
> 06.03.12 12:40:41.287 EP(2): INTERFACE (no free ports found)
> 06.03.12 12:40:41.287 EP(1): TONE to CH(1) directory default name
> cause_22
> 06.03.12 12:40:41.287 EP(1): DISCONNECT to CH(1) cause value=34
> location=1-Local-PBX
> 06.03.12 12:40:41.287 CH(1): MNCC_DISC_REQ LCR<->BSC progress coding=3
> location=1 descr=8 cause coding=3 location=1 value=34
> 06.03.12 12:40:56.246 CH(1): MNCC_REL_IND LCR<->BSC cause coding=3
> location=0 value=16
> 06.03.12 12:40:56.247 EP(1): RELEASE from CH(1) cause value=16
> location=0-User
> 06.03.12 12:40:56.247 EP(1): ACTION hangup
>
>
> On Tue, Mar 6, 2012 at 7:44 AM, Nik Pakar <nikpakar(a)gmail.com> wrote:
>>
>> Hi Andreas,
>>
>> It was apperently compiled on my debian while i had misdn libs
>> isntalled. Now im trying on a fresh debian from the same set of sources
>> which i got it working and without misdn, it fails to compile gsm.
>>
>> Attached is my config output and compile break.
>>
>> http://pastebin.com/W6UHn4Lc
>>
>> So should i still install misdn even though its not used.
>>
>> Rgds
>> Nik
>>
>>
>> On Mon, Mar 5, 2012 at 10:05 AM, Andreas Eversberg
>> <andreas(a)eversberg.eu> wrote:
>>>
>>> Alexander Chemeris wrote:
>>> > Does it mean that that now we can use LCR with other SIP
>>> > softswitches/PBX'es, like Freeswitch? I do not follow LCR
>>> > development
>>> > closely, but that would be a very interesting development.
>>> >
>>> yes, this was my intention. gsm and sip interface of lcr will not
>>> rely
>>> on misdn anymore. currently i don't support audio transfer via
>>> chan_lcr,
>>> so chan_lcr will only work with isdn interfaces. the sip interface
>>> implementation has not much options, so it can only do sipmple
>>> point-to-point sip connections to a gateway or endpoint.
>>
>>
>