Dear SERusers and SERdevelopers, ------------------------------------- For geographic information system we want to use SIP Voip over satellite. The attached figure will give you a short overview of the network architecture. We have a home network and a field network. The field network is directly behind the satellite station. In the field network we will have notebooks and PDA running Voip SIP client software. The VSAT and the WLAN Accesspoint will be mounted in a VAN or other movable vehicle. In the VAN a operation team will control all the proccesses in the field network. For Voip online collaboration they have several Voip hardphones. We'd like to havve Voip bidirectional connection between the operating team and the stuff in the field and Voip conferencing as well. For emergency situations e.g. the field network is a mine field of a demining NGO the operating team should be able to send a emergency call to everybody in the mine filed to stop the demining proccess immediately. Right now demining organizations use VHF radio for this purpose. The operation team in the home office should be able to call everybody in the filed network and participate in the field's Voip conference.
To test the Voip functionality I set up a demonstrator network and installed the developement versions of SER and SEMS in the home network and in the field network. Because I am a new in this realm I have additional questions the administrator manual could not answer or I did not unserstand it.
1. Do I need DNS running for SER routing between the home and field network even I am using privat networks ?
2. How and were can I define a conference URI e.g. sip:fieldconf@10.3.240.1
3. The postgresql SER module does not compile under RH 7.3 and RH 9.0 . I cannot find a SQL setup script for postresql. Do I really need a RDMS like MySQL for user management and registration?
4. Does the following scriptlet routes all matching calls from the home network to the field network. ( All numbers beginning with a 2 should be routed to the filed network and all URI'S containing 10.3.240.1 in the domain part:
if(uri=~"^sip:2[0-9][0-9][0-9] | (10.3.240.1)" { forward (10.3.240.1, 5060); }
All extension beginning with 1 or containing 10.3.1.2 as domain part in the URI should be routed to the SER in the home network.
????
Thank you for your help
Olaf
-- Fhg-Fokus-CC Satcom
Hello, comments inline.
On 10-11 12:07, Olaf Menzel wrote:
Dear SERusers and SERdevelopers,
For geographic information system we want to use SIP Voip over satellite. The attached figure will give you a short overview of the network architecture. We have a home network and a field network. The field network is directly behind the satellite station. In the field network we will have notebooks and PDA running Voip SIP client software. The VSAT and the WLAN Accesspoint
May I know what client software are you going to use ?
will be mounted in a VAN or other movable vehicle. In the VAN a operation team will control all the proccesses in the field network. For Voip online collaboration they have several Voip hardphones. We'd like to havve Voip bidirectional connection between the operating team and the stuff in the field and Voip conferencing as well. For emergency situations e.g. the field network is a mine field of a demining NGO the operating team should be able to send a emergency call to everybody in the mine filed to stop the demining proccess immediately. Right now demining organizations use VHF radio for this purpose. The operation team in the home office should be able to call everybody in the filed network and participate in the field's Voip conference.
To test the Voip functionality I set up a demonstrator network and installed the developement versions of SER and SEMS in the home network and in the field network. Because I am a new in this realm I have additional questions the administrator manual could not answer or I did not unserstand it.
- Do I need DNS running for SER routing between the home and field
network even I am using privat networks ?
No, for routing between home and field network you don't need DNS. You can create static routes in the configuration script using IP. Note that you will also probably need SIP devices that can operate without DNS, most SIP devices available today resolve the server's IP address using SRV records.
You can simply forward all the SIP traffic from the field network which is not targeted to the field network to your home network server.
- How and were can I define a conference URI e.g. sip:fieldconf@10.3.240.1
In the configuration script of ser. If you detect a conference URI then you will forward it to the conference server.
- The postgresql SER module does not compile under RH 7.3 and RH 9.0 .
Any error message ?
I cannot find a SQL setup script for postresql.
Currently we have only ser_mysql.sh which works with MySQL only. A version for postgres currently doesn't exist, but we plan to introduce it in the future.
Do I really need a RDMS like MySQL for user management andregistration?
No, you can use dbtext as well which can store all the data in plaintext files. It is not suitable for a large amount of data, but for smaller setups like yours it should work fine.
- Does the following scriptlet routes all matching calls from the home
network to the field network. ( All numbers beginning with a 2 should be routed to the filed network and all URI'S containing 10.3.240.1 in the domain part:
if(uri=~"^sip:2[0-9][0-9][0-9] | (10\.3\.240\.1)" { forward (10.3.240.1, 5060); }
The condition will be true for all SIP URIs that begin with 2 followed by at least 3 digits or SIP URI that contain "10.3.240.1".
Probably you want something like this:
if (uri =~ "(^sip:2[0-9]*@.*)|(^sip:.*@10.3.240.1)")
Also use t_relay_to_udp instead of forward. Forward implements stateless forwarding while t_relay_to_udp implements stateful forwarding.
All extension beginning with 1 or containing 10.3.1.2 as domainpart in the URI should be routed to the SER in the home network.
Similar to the previous condition.
Jan.
Jan Janak wrote:
May I know what client software are you going to use ?
We are using SJPhone For Windows 2000 and PocketPC and TkcPhone for the Sharp Zaurus. For native Linux we are playing aaround with kphone, linphone, sipset, tkphone, ... but did not get suitable audio quality. We are still waiting for SJ LAbs Linux version. At leatest we'd like to substitute all of the mentioned clients by a Java Client to be integrate it into the already developed J2ME geo-client. What about the Sip communicator (https://sip-communicator.dev.java.net/) ? Do you have any expierences ?
- The postgresql SER module does not compile under RH 7.3 and RH 9.0 .
Any error message ?
make[1]: Entering directory `/usr/src/sip_router/modules/postgres' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -m486 -malign-loops=4 -DNAME='"ser"' -DVERSION='"0.8.12dev-22-tcp_aliases"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 2.96"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -I/usr/include -I/usr/include/pgsql -c db_mod.c -o db_mod.o db_mod.c:51: `use_table' undeclared here (not in a function) db_mod.c:51: initializer element is not constant db_mod.c:51: (near initialization for `cmds[0].function') make[1]: *** [db_mod.o] Error 1 make[1]: Leaving directory `/usr/src/sip_router/modules/postgres'
Thank you
Olaf
On 10-11 21:08, Olaf Menzel wrote:
Jan Janak wrote:
May I know what client software are you going to use ?
We are using SJPhone For Windows 2000 and PocketPC and TkcPhone for the Sharp Zaurus. For native Linux we are playing aaround with kphone, linphone, sipset, tkphone, ... but did not get suitable audio quality. We are still waiting for SJ LAbs Linux version. At leatest we'd like to substitute all of the mentioned clients by a Java Client to be integrate it into the already developed J2ME geo-client. What about the Sip communicator (https://sip-communicator.dev.java.net/) ? Do you have any expierences ?
No.
- The postgresql SER module does not compile under RH 7.3 and RH 9.0 .
Any error message ?
make[1]: Entering directory `/usr/src/sip_router/modules/postgres' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -m486 -malign-loops=4 -DNAME='"ser"' -DVERSION='"0.8.12dev-22-tcp_aliases"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 2.96"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DDNS_IP_HACK -DUSE_IPV6 -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -I/usr/include -I/usr/include/pgsql -c db_mod.c -o db_mod.o db_mod.c:51: `use_table' undeclared here (not in a function) db_mod.c:51: initializer element is not constant db_mod.c:51: (near initialization for `cmds[0].function') make[1]: *** [db_mod.o] Error 1 make[1]: Leaving directory `/usr/src/sip_router/modules/postgres'
Are you using unstable ser ? It looks like you are using a version with my recent database changes.
Jan.
At 09:08 PM 11/10/2003, Olaf Menzel wrote:
We are using SJPhone For Windows 2000 and PocketPC and TkcPhone for the Sharp Zaurus.
xten could do a great job, it supports the very nice gips/ilbc codec.
For native Linux we are playing aaround with kphone, linphone, sipset, tkphone, ... but did not get suitable audio quality. We are still waiting for SJ LAbs Linux version. At leatest we'd like to substitute all of the mentioned clients by a Java Client to be integrate it into the already developed J2ME geo-client. What about the Sip communicator (https://sip-communicator.dev.java.net/) ? Do you have any expierences ?
what is java?
-jiri
Jiri Kuthan wrote:
At 09:08 PM 11/10/2003, Olaf Menzel wrote:
We are using SJPhone For Windows 2000 and PocketPC and TkcPhone for the Sharp Zaurus.
xten could do a great job, it supports the very nice gips/ilbc codec.
I forgot TheKompany's TkPhone for Linux
For native Linux we are playing aaround with kphone, linphone, sipset, tkphone, ... but did not get suitable audio quality. We are still waiting for SJ LAbs Linux version. At leatest we'd like to substitute all of the mentioned clients by a Java Client to be integrate it into the already developed J2ME geo-client. What about the Sip communicator (https://sip-communicator.dev.java.net/) ? Do you have any expierences ?
what is java?
Mabye Google helps ;-)
There are 2 additional projects for JAVA SIP implementation:
- One of our students at FH Rhein-Sieg is working on it at: http://devel.stud.inf.fh-bonn-rhein-sieg.de/cvs/?group_id=50
- Nist has developed its Jain-SIP API and some apps based on this API: http://snad.ncsl.nist.gov/proj/iptel/
Olaf
Hi Jiri, Jan, Raphael and all other Ser users and developers, --------------------------------------------------------------
as I have already mentioned the demining scenario in our project there is the need of a Voip SIP "broadcast" functionality. This means the resonsable operator should be ready to reach the whole demining group with one call. Because we do not have real braodcast like VHF radio a approach could be the following scenario:
The operator will send a SIP invite to all members of his team and when the user ansers the phone call the call will automaticly transfered to a SIP conference. This is what I have discussed with Raphael. What do you think how to realize such a "broadcast" functionality ??
regards
Olaf
Hello,
please don't use both serusers@lists.iptel.org and serdev@lists.iptel.org. Pick one list only. I would recommend serusers in this case since it is not about the development of the proxy.
I agree to the conferencing stuff, you could use REFER to do the transfer.
Jan.
On 11-11 13:45, Olaf Menzel wrote:
Hi Jiri, Jan, Raphael and all other Ser users and developers,
as I have already mentioned the demining scenario in our project there is the need of a Voip SIP "broadcast" functionality. This means the resonsable operator should be ready to reach the whole demining group with one call. Because we do not have real braodcast like VHF radio a approach could be the following scenario:
The operator will send a SIP invite to all members of his team and when the user ansers the phone call the call will automaticly transfered to a SIP conference. This is what I have discussed with Raphael. What do you think how to realize such a "broadcast" functionality ??
regards
Olaf
Serdev mailing list serdev@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serdev
There are two parts: one of them is signaling, the other media. MEdia is delt with using a conferencing bridge, that's it. What is left then is a signaling fan.
I argue that the fan functionality is better left away from SEMS and put in a specialized software component. (specialization is good). I would compare it to mailman which takes care of groups and runs separately from sendmail.
The call flow would be that the group management software instructs SEMS using REFERs to invite all parties in question to a conference. I.e., SEMS acts in dial-out manner. The advantage over sending REFER to participants is it works with REFER-less telephones too.
-jiri
At 01:45 PM 11/11/2003, Olaf Menzel wrote:
Hi Jiri, Jan, Raphael and all other Ser users and developers,
as I have already mentioned the demining scenario in our project there is the need of a Voip SIP "broadcast" functionality. This means the resonsable operator should be ready to reach the whole demining group with one call. Because we do not have real braodcast like VHF radio a approach could be the following scenario:
The operator will send a SIP invite to all members of his team and when the user ansers the phone call the call will automaticly transfered to a SIP conference. This is what I have discussed with Raphael. What do you think how to realize such a "broadcast" functionality ??
regards
Olaf
Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers
-- Jiri Kuthan http://iptel.org/~jiri/
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Jan Janak -
Jiri Kuthan -
Olaf Menzel