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Nik Pakar nikpakar at gmail.comHi Alexander, I will check that tomorrow and update. But the problem here is LCR get GSM-FR from MNCC and forward SDP as PCMA on SIP. So it will confuse the softswitch as it see the stream as PCMA even it comes on FR. and if its actually PCMA, then LCR must have transcode it. Either way no audio once call connected. On openbsc console i see some warning on lchan with no rtp. Rgds Nik On Wed, Mar 7, 2012 at 6:32 PM, Alexander Chemeris < alexander.chemeris at gmail.com> wrote: > 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 at 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 at 192.168.1.30 to > > uri=sip:4290080001 at 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 at 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 at 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 at 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 at 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 at 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 at 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. > -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.osmocom.org/pipermail/openbsc/attachments/20120307/6734644d/attachment.htm>