Hi Alexander,
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@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@gmail.com> wrote:
>>
>> 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@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@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@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@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@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.
>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
Regards,
Alexander Chemeris.