Hello list, First of all congratulation to developers of this project making such wonderful project in an interesting subject. I am new to this project and for past few days i managed to go through the documentation and get most of it tested. Thanks for pretty good documentation in most of the areas as well. I have the osmo-nitb working very fine using a nano bts. How ever when i try to install it with LCR to interconnect with external switch, im facing some problems. Initially i thought i must use LCR+Asterisk. But later i figured out there is a built in SIP interface on LCR which there is no need to asterisk or chan_asterisk. I would prefer to use this LCR SIP interface as i dont want to use asterisk and just want to forward all calls to another SIP switch. Now in this context there seems absolutely no documentation on both openbsc and LCR/mISDN lists. Can some one please shed me some light here on how to build a LCR with SIP to be work with osmo-nitb. All i want to test is GSM phone > Osmo-nitb > LCR with SIP > SIP softswitch Thank you very much for every one's effort in this project and would be glad to see some response for this. Best Regards Nava.
Hi Nava, I have tried the same some time ago. LCR-SIP implementation seems very nice in what you trying to do without using asterisk. I managed to get calls ring back and forward in a similar setup but i was running into serious problems with RTP/media. This was mainly due to transcoding or rtp-bridge functions. I was some what successful with the help from andreas but could not make it work 100%. May be now there is a fix for the rtp issues which i never tried after that. Give it a try using the latest git clone from LCR and see. As i can remember you need to check out jolly/rtpmux branch in order to get this working. Andreas can give some hint if possible on this. Good luck. Nik. On Mon, Oct 15, 2012 at 9:54 PM, Kelum Navaratne <kelum.nava@gmail.com>wrote:
Hello list,
First of all congratulation to developers of this project making such wonderful project in an interesting subject. I am new to this project and for past few days i managed to go through the documentation and get most of it tested. Thanks for pretty good documentation in most of the areas as well.
I have the osmo-nitb working very fine using a nano bts.
How ever when i try to install it with LCR to interconnect with external switch, im facing some problems. Initially i thought i must use LCR+Asterisk. But later i figured out there is a built in SIP interface on LCR which there is no need to asterisk or chan_asterisk. I would prefer to use this LCR SIP interface as i dont want to use asterisk and just want to forward all calls to another SIP switch.
Now in this context there seems absolutely no documentation on both openbsc and LCR/mISDN lists.
Can some one please shed me some light here on how to build a LCR with SIP to be work with osmo-nitb.
All i want to test is
GSM phone > Osmo-nitb > LCR with SIP > SIP softswitch
Thank you very much for every one's effort in this project and would be glad to see some response for this.
Best Regards Nava.
hi kelum, in order to use sip, you need jolly/new branch of lcr and jolly/rtpmux of openbsc. the sip implementation of LCR is quite simple, so no authentication or oder features - just simple point-to-point SIP. if you run confiure of LCR, check if sip is enabled. in order to add a SIP interface, do the following at interface.conf: [sip] sip <local ip>[:<local port>] <remote ip>[:<remote port>] earlyb yes tones no you need to define local IP that will be used to connect to remote SIP endpoint. don't use localhost, if the endpoint is on a different machine, because this IP is also used for RTP. if you use same machine, you need to have different ports. you may change local port, by adding a local port or you may change port of SIP endpoint and then add remote port. i have tested it with freeswitch, but asterisk should work also. you may then also try at interface.conf below "[sip]" definition: rtp-brige then the codec (full rate or enhanced full rate) is negotiated between mobile and the remote SIP endpoint. the SIP endpoint must support at least one of it. regards, andreas
Hi Andreas, Thank you very much. This is a good start for me. I also need to work with freeswitch. Not asterisk. And simple point to point sip is more than enough and rtp bridge would be fantastic. Would it work with amr codec as well ? Or only gsm full rate/ h. Rate? Im trying to find the lcr git location of jolly/new. Can you please give me the url. I cant find it on openbsc git. Thanks again. Nava. On Tuesday, October 16, 2012, Andreas Eversberg wrote:
hi kelum,
in order to use sip, you need jolly/new branch of lcr and jolly/rtpmux of openbsc. the sip implementation of LCR is quite simple, so no authentication or oder features - just simple point-to-point SIP. if you run confiure of LCR, check if sip is enabled. in order to add a SIP interface, do the following at interface.conf:
[sip] sip <local ip>[:<local port>] <remote ip>[:<remote port>] earlyb yes tones no
you need to define local IP that will be used to connect to remote SIP endpoint. don't use localhost, if the endpoint is on a different machine, because this IP is also used for RTP. if you use same machine, you need to have different ports. you may change local port, by adding a local port or you may change port of SIP endpoint and then add remote port.
i have tested it with freeswitch, but asterisk should work also.
you may then also try at interface.conf below "[sip]" definition:
rtp-brige
then the codec (full rate or enhanced full rate) is negotiated between mobile and the remote SIP endpoint. the SIP endpoint must support at least one of it.
regards,
andreas
Hi Every one, Would the IP.Access nanoBTS works behind a NAT with OpenBSC ? i.e. two BTSs in two different locations behind NAT connecting to NITB. im trying to make a call, but it seems rtp has some problem. rgds Nava On Tue, Oct 16, 2012 at 6:49 PM, Kelum Navaratne <kelum.nava@gmail.com>wrote:
Hi Andreas,
Thank you very much. This is a good start for me. I also need to work with freeswitch. Not asterisk. And simple point to point sip is more than enough and rtp bridge would be fantastic.
Would it work with amr codec as well ? Or only gsm full rate/ h. Rate?
Im trying to find the lcr git location of jolly/new. Can you please give me the url. I cant find it on openbsc git.
Thanks again.
Nava.
On Tuesday, October 16, 2012, Andreas Eversberg wrote:
hi kelum,
in order to use sip, you need jolly/new branch of lcr and jolly/rtpmux of openbsc. the sip implementation of LCR is quite simple, so no authentication or oder features - just simple point-to-point SIP. if you run confiure of LCR, check if sip is enabled. in order to add a SIP interface, do the following at interface.conf:
[sip] sip <local ip>[:<local port>] <remote ip>[:<remote port>] earlyb yes tones no
you need to define local IP that will be used to connect to remote SIP endpoint. don't use localhost, if the endpoint is on a different machine, because this IP is also used for RTP. if you use same machine, you need to have different ports. you may change local port, by adding a local port or you may change port of SIP endpoint and then add remote port.
i have tested it with freeswitch, but asterisk should work also.
you may then also try at interface.conf below "[sip]" definition:
rtp-brige
then the codec (full rate or enhanced full rate) is negotiated between mobile and the remote SIP endpoint. the SIP endpoint must support at least one of it.
regards,
andreas
Kelum Navaratne wrote:
Would the IP.Access nanoBTS works behind a NAT with OpenBSC ?
i.e. two BTSs in two different locations behind NAT connecting to NITB. im trying to make a call, but it seems rtp has some problem. hi kelum,
the NAT firewall should track the RTP frames from BTS behind NAT, so that frames from BSC outside NAT can find their way to the BTS. i suggest to sniff (tcpdump/wireshark) behind and before NAT, in order to localize the problem. regards, andreas
Hi Andreas, this is this mean a BTS behind NAT and a OpenBSC +LCR/freeswitch + sip phone on freeswitch will be able to communicate with good RTP ? Rgds Nik On Wed, Jan 23, 2013 at 1:11 PM, Andreas Eversberg <andreas@eversberg.eu>wrote:
Kelum Navaratne wrote:
Would the IP.Access nanoBTS works behind a NAT with OpenBSC ?
i.e. two BTSs in two different locations behind NAT connecting to NITB. im trying to make a call, but it seems rtp has some problem.
hi kelum,
the NAT firewall should track the RTP frames from BTS behind NAT, so that frames from BSC outside NAT can find their way to the BTS. i suggest to sniff (tcpdump/wireshark) behind and before NAT, in order to localize the problem.
regards,
andreas
Andreas Eversberg wrote:
the NAT firewall should track the RTP frames
Nik Pakar wrote:
this is this mean a BTS behind NAT and a OpenBSC +LCR/freeswitch + sip phone on freeswitch will be able to communicate with good RTP ?
I think Andreas wrote pretty clearly that it depends on if the NAT firewall can and will track RTP or not; it has nothing to do with the software on either side. Follow Andreas' advice and experiment with different configurations while studying the packets on both sides of the firewall. //Peter
Hello andreas, Im trying to get the jolly/new branch of lcr, but there is no such branch on the tree. below is my output. can you please let me know whether im checking out from the right repo. root@debian:/software/lcr# git clone git://git.misdn.eu/lcr.git/ Cloning into lcr... remote: Counting objects: 3695, done. remote: Compressing objects: 100% (1328/1328), done. remote: Total 3695 (delta 2716), reused 3226 (delta 2353) Receiving objects: 100% (3695/3695), 5.76 MiB | 8.27 MiB/s, done. Resolving deltas: 100% (2716/2716), done. root@debian:/software/lcr/lcr# git branch -a * master remotes/origin/1.10 remotes/origin/1.11 remotes/origin/1.13 remotes/origin/1.8 remotes/origin/1.9 remotes/origin/HEAD -> origin/master remotes/origin/holger/cleaning remotes/origin/holger/cleaning-rebased remotes/origin/jolly/vootp remotes/origin/master remotes/origin/pending root@debian:/software/lcr/lcr# On Tue, Oct 16, 2012 at 1:51 PM, Andreas Eversberg <andreas@eversberg.eu>wrote:
hi kelum,
in order to use sip, you need jolly/new branch of lcr and jolly/rtpmux of openbsc. the sip implementation of LCR is quite simple, so no authentication or oder features - just simple point-to-point SIP. if you run confiure of LCR, check if sip is enabled. in order to add a SIP interface, do the following at interface.conf:
[sip] sip <local ip>[:<local port>] <remote ip>[:<remote port>] earlyb yes tones no
you need to define local IP that will be used to connect to remote SIP endpoint. don't use localhost, if the endpoint is on a different machine, because this IP is also used for RTP. if you use same machine, you need to have different ports. you may change local port, by adding a local port or you may change port of SIP endpoint and then add remote port.
i have tested it with freeswitch, but asterisk should work also.
you may then also try at interface.conf below "[sip]" definition:
rtp-brige
then the codec (full rate or enhanced full rate) is negotiated between mobile and the remote SIP endpoint. the SIP endpoint must support at least one of it.
regards,
andreas
Kelum Navaratne wrote:
Im trying to get the jolly/new branch of lcr
Seriously? Come on..
On Tue, Oct 16, 2012 at 1:51 PM, Andreas Eversberg <andreas@eversberg.eu> wrote:
in order to use sip, you need jolly/new branch of lcr
Guess if there was some activity in the repository in the last three months? Oh wait - this is open source, so actually you don't have to guess. You can use the git log command to study the commit history and IMMEDIATELY see what development has happened in the repository. I suggest that you try that now. If you are more ambitious then you will of course additionally read the git-log manual, where you will find that you can use various methods to search the repository history. You might come up with a command like: git log --all -i --grep=rtp ..which you can follow up with a command like: git branch --contains insert_commit_id_here ..to learn which branches contain the most interesting commits. I advise you to study git, so that you will be able to use source code managed in git repositories. If you don't know the tools then you're very limited, and the OpenBSC community is obviously the wrong place for git training. There are plenty of resources online. I recommend studying the free-as-in-beer book called Pro Git. Ask google.com about it. //Peter
Kelum Navaratne wrote:
Im trying to get the jolly/new branch of lcr, but there is no such branch on the tree. below is my output. can you please let me know whether im checking out from the right repo. hi kelum,
it's gone - deleted. because it's already merged with master. regards, andreas
participants (4)
-
Andreas Eversberg -
Kelum Navaratne -
Nik Pakar -
Peter Stuge