Re: [OpenSIPS-Users] What is the module to fix SDP ?
Hi Alex. The project I´m currently working on was requested by a customer from the telecommunication market. I don´t know the business details as my manager, but what I know is : -the entire system has to run in a ARM hardware. It is an embedded system, with limited hardware resources. -due to custom reasons, it was decided not to use another hardware to run media relay. That is why they have decided to implement direct media. Yes I know. Thank you very much for talking with me about this ideas. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 Brazil De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Alex Balashov <abalas...@evaristesys.com> Enviado: quinta-feira, 31 de março de 2016 15:15 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] What is the module to fix SDP ? As long as the STUN client properly ascertains its external RTP port through STUN as well, that should work fine. Out of curiosity, what are your hardware limitations that would make server-side RTP relay impractical? Also, you are aware that your RTP relay doesn't have to be on the same hardware as your SIP proxy, correct? -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is the module to fix SDP ?
As long as the STUN client properly ascertains its external RTP port through STUN as well, that should work fine. Out of curiosity, what are your hardware limitations that would make server-side RTP relay impractical? Also, you are aware that your RTP relay doesn't have to be on the same hardware as your SIP proxy, correct? -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is the module to fix SDP ?
Hi Alex. That is the point. We have just decided what to do: We will have a STUN server in the same hardware where is OpenSIPS and a softphone (this softphone will talk to others softphones from Internet). Such hardware will be connected to a switch and such switch will have a public IP. The switch will work as a NAT. OpenSIPS and the main softphone will be behind this NAT. So, when the main softphone communicates with other one from Internet, this one will have to fix the SDP (IP and PORT) so that the one from Internet will be able to send its media to the right end point. This work of fixing SDP will use the STUN + ICE. Our intention is to implement direct media. We cannot implement a media relay in our hardware because it doesn't have sufficient resources for this kind of relay. Tell me if it is clear and if it sounds good, please. Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome de Alex Balashov <abalas...@evaristesys.com> Enviado: quinta-feira, 31 de março de 2016 14:31 Para: users@lists.opensips.org Assunto: Re: [OpenSIPS-Users] What is the module to fix SDP ? Hello Rodrigo, If you're not going to use an outboard media relay, how do you propose to "fix" media ports in the SDP offers & answers? What will you rewrite them to if you have no RTP endpoint to give you a means of seeing where media is coming from? :-) -- Alex -- Alex Balashov | Principal | Evariste Systems LLC 1447 Peachtree Street NE, Suite 700 Atlanta, GA 30309 United States Tel: +1-800-250-5920 (toll-free) / +1-678-954-0671 (direct) Web: http://www.evaristesys.com/, http://www.csrpswitch.com/ ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Re: [OpenSIPS-Users] What is the module to fix SDP ?
Hi Carlos.
That is what I'm looking for and now trying to understand if it is sufficient
for may case.
With this function I will be able to fix IP. But I still have to fix Ports.
If there are NATs, so I suppose there is some kind of Ports mapping that has to
be taken into account when solving the NAT traversal. I and a coworker are
investigating how to solve the port mapping. Maybe the solution will demand ICE
and STUN. We can't use media proxy module, because our hardware will not be
able to do media relay. So, We can have a media relay in the same place as
OpenSIPs.
Thanks a lot.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200 RAMAL 979
De: users-boun...@lists.opensips.org <users-boun...@lists.opensips.org> em nome
de Carlos Eduardo <kad...@gmail.com>
Enviado: quinta-feira, 31 de março de 2016 11:43
Para: OpenSIPS users mailling list
Assunto: Re: [OpenSIPS-Users] What is the module to fix SDP ?
Hello Rodrigo,
Maybe the function fix_nated_sdp, from nathelper module, would help.
1.5.2. fix_nated_sdp(flags [, ip_address])
Alters the SDP information in orer to facilitate NAT traversal. What changes to
be performed may be controled via the "flags" paramter.
Meaning of the parameters is as follows:
flags - the value may be a bitwise OR of the following flags:
0x01 - adds "a=direction:active" SDP line;
0x02 - rewrite media IP address (c=) with source address of the message or the
provided IP address (the provide IP address take precedence over the source
address).
0x04 - adds "a=nortpproxy:yes" SDP line;
0x08 - rewrite IP from origin description (o=) with source address of the
message or the provided IP address (the provide IP address take precedence over
the source address).
ip_address - IP to be used for rewriting SDP. If not specified, the received
signalling IP will be used. The parameter allows pseudo-variables usage. NOTE:
For the IP to be used, you need to use 0x02 or 0x08 flags, otherwise it will
have no effect.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE,
BRANCH_ROUTE.
Example:
if (search("User-Agent: Cisco ATA.*") {fix_nated_sdp("3");};
2016-03-30 14:57 GMT-03:00 Rodrigo Pimenta Carvalho
<pime...@inatel.br<mailto:pime...@inatel.br>>:
Hi.
I have a case when a peer answering a call (INVITE) is behind a NAT.
So, in its SIP OK message I would like to see the SDP containing a valid IP and
media Port valid to receive audio from the caller. That is, the caller need to
know a valid IP and Port where he/she can send his/her audio packets.
1 - Is it possible to "fix" SDP content for such objective?
2 - Can OpenSIPS do something for this idea works or must I to use something
more like a stun server?
3 - What is the OpenSIPS module that can help me with this task?
I guess I will have to fix 2 fiels in SDP:
Media Description, name and address (m): audio 55142 RTP/AVP 8 101 ( to
fix the port)
Connection Information (c): IN IP4 192.168.100.156
( to fix the IP)
P.S.: I have already a solution (opensips.cfg) that let SIP messages cross NATs
without problems. Only SDP has to be fixed.
Any hint will be very helpful!
Best regards.
RODRIGO PIMENTA CARVALHO
Inatel Competence Center
Software
Ph: +55 35 3471 9200<tel:%2B55%2035%203471%209200> RAMAL 979
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--
Carlos E. Wagner
Tecnólogo em Telecomunicações, OCP, dCAA
Gnotel Tecnologia
E-mail: kad...@gmail.com<mailto:kad...@gmail.com>
Fone: +55 48 9981-0894
Skype: carlos.e.wagner
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Re: [OpenSIPS-Users] What is the module to fix SDP ?
Hello Rodrigo,
Maybe the function fix_nated_sdp, from nathelper module, would help.
1.5.2. fix_nated_sdp(flags [, ip_address])
Alters the SDP information in orer to facilitate NAT traversal. What
changes to be performed may be controled via the “flags” paramter.
Meaning of the parameters is as follows:
flags - the value may be a bitwise OR of the following flags:
0x01 - adds “a=direction:active” SDP line;
0x02 - rewrite media IP address (c=) with source address of the message or
the provided IP address (the provide IP address take precedence over the
source address).
0x04 - adds “a=nortpproxy:yes” SDP line;
0x08 - rewrite IP from origin description (o=) with source address of the
message or the provided IP address (the provide IP address take precedence
over the source address).
ip_address - IP to be used for rewriting SDP. If not specified, the
received signalling IP will be used. The parameter allows pseudo-variables
usage. NOTE: For the IP to be used, you need to use 0x02 or 0x08 flags,
otherwise it will have no effect.
This function can be used from REQUEST_ROUTE, ONREPLY_ROUTE, FAILURE_ROUTE,
BRANCH_ROUTE.
Example:
if (search("User-Agent: Cisco ATA.*") {fix_nated_sdp("3");};
2016-03-30 14:57 GMT-03:00 Rodrigo Pimenta Carvalho :
> Hi.
>
>
> I have a case when a peer answering a call (INVITE) is behind a NAT.
>
> So, in its SIP OK message I would like to see the SDP containing a valid
> IP and media Port valid to receive audio from the caller. That is, the
> caller need to know a valid IP and Port where he/she can send his/her audio
> packets.
>
>
> 1 - Is it possible to "fix" SDP content for such objective?
>
> 2 - Can OpenSIPS do something for this idea works or must I to use
> something more like a stun server?
>
> 3 - What is the OpenSIPS module that can help me with this task?
>
>
> I guess I will have to fix 2 fiels in SDP:
>
> Media Description, name and address (m): audio 55142 RTP/AVP 8 101 (
> to fix the port)
> Connection Information (c): IN IP4 192.168.100.156
> ( to fix the IP)
>
> P.S.: I have already a solution (opensips.cfg) that let SIP messages cross
> NATs without problems. Only SDP has to be fixed.
>
>
> Any hint will be very helpful!
>
> Best regards.
>
>
>
> RODRIGO PIMENTA CARVALHO
> Inatel Competence Center
> Software
> Ph: +55 35 3471 9200 RAMAL 979
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
--
*Carlos E. Wagner*
*Tecnólogo em Telecomunicações, OCP, dCAA*
*Gnotel Tecnologia*
*E-mail:* *kad...@gmail.com *
*Fone:* +55 48 9981-0894
*Skype:* carlos.e.wagner
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[OpenSIPS-Users] What is the module to fix SDP ?
Hi. I have a case when a peer answering a call (INVITE) is behind a NAT. So, in its SIP OK message I would like to see the SDP containing a valid IP and media Port valid to receive audio from the caller. That is, the caller need to know a valid IP and Port where he/she can send his/her audio packets. 1 - Is it possible to "fix" SDP content for such objective? 2 - Can OpenSIPS do something for this idea works or must I to use something more like a stun server? 3 - What is the OpenSIPS module that can help me with this task? I guess I will have to fix 2 fiels in SDP: Media Description, name and address (m): audio 55142 RTP/AVP 8 101 ( to fix the port) Connection Information (c): IN IP4 192.168.100.156 ( to fix the IP) P.S.: I have already a solution (opensips.cfg) that let SIP messages cross NATs without problems. Only SDP has to be fixed. Any hint will be very helpful! Best regards. RODRIGO PIMENTA CARVALHO Inatel Competence Center Software Ph: +55 35 3471 9200 RAMAL 979 ___ Users mailing list Users@lists.opensips.org http://lists.opensips.org/cgi-bin/mailman/listinfo/users
