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sridhart at alcatel-lu... Guest
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Posted: Fri May 22, 2009 12:28 am Post subject: [Freeswitch-users] Help regarding configuration of FreeSWITC |
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Hi all,
I want to use FreeSWITCH as a SIP transparent proxy in session border controller application. Please let me know the changes in configuration files required to achieve this behaviour
Thanks very much for the help.
Regards,
Sridhar |
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oseslija at gmail.com Guest
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Posted: Fri May 22, 2009 5:24 am Post subject: [Freeswitch-users] Help regarding configuration of FreeSWITC |
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Hello,
FS by design is B2BUA, and it cannot route INVITEs and other SIP methods. It can however, bridge a-leg to b-leg with or w/o media and doing plenty other cool stuff much better than commercial projects. I suggest joining us on irc to detail your setup so we can help you.
Regards,
Ognjen (sekil on #freeswitch).
On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar) <sridhart@alcatel-lucent.com (sridhart@alcatel-lucent.com)> wrote:
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Prometheus001 at gmx.net Guest
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Posted: Fri May 22, 2009 7:53 am Post subject: [Freeswitch-users] Help regarding configuration of FreeSWITC |
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This is also interesting for me, as I love freeswitch, and maintaining a
single platform is easier, than handling various different ones.
In the past years I did a couple of projects with OpenSER /openSIPS.
These projects comprised:
* registrar for the SIP user agents
* handle invite messages (+ ringing, bye, ok, etc also of course)
between registered user agents and user agents at external domains
* rtp payload was a bit different from usual VoIP traffic (video
parts, application sharing, file downloads etc.), but SDP was fine
according to RFC, and OpenSER mediaproxy worked also
* handling of peer-to-peer presence (SUBSCRIBE, MEASSAGE, OPTIONS)
* The number of messages to handle was not that much (some thousand
subs).
For my understanding this should also be possible with Freeswitch with
bypass_media. Right?
Best regards
Peter
Ognjen Seslija schrieb:
Quote: | Hello,
FS by design is B2BUA, and it cannot route INVITEs and other SIP
methods. It can however, bridge a-leg to b-leg with or w/o media and
doing plenty other cool stuff much better than commercial projects. I
suggest joining us on irc to detail your setup so we can help you.
Regards,
Ognjen (sekil on #freeswitch).
On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar)
<sridhart@alcatel-lucent.com <mailto:sridhart@alcatel-lucent.com>> wrote:
Hi all,
I want to use FreeSWITCH as a SIP transparent proxy in session
border controller application. Please let me know the changes in
configuration files required to achieve this behaviour
Thanks very much for the help.
Regards,
Sridhar
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oseslija at gmail.com Guest
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Posted: Sat May 23, 2009 2:49 am Post subject: [Freeswitch-users] Help regarding configuration of FreeSWITC |
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It can be done with or without media. FS does not relay SIP messages transparently as OpenSER, but act as a SIP UA to each call leg. This gives an oportunity to place rtp in between streams, do call recording and all other softswitch stuff. I use FreeSWITCH as hosted PBX solution, basically hosting multiple SIP domains each of them representing a separate PBX for a different company with separate dialplans etc.
Ognjen
On Fri, May 22, 2009 at 2:37 PM, Peter P GMX <Prometheus001@gmx.net (Prometheus001@gmx.net)> wrote:
Quote: | This is also interesting for me, as I love freeswitch, and maintaining a
single platform is easier, than handling various different ones.
In the past years I did a couple of projects with OpenSER /openSIPS.
These projects comprised:
* registrar for the SIP user agents
* handle invite messages (+ ringing, bye, ok, etc also of course)
between registered user agents and user agents at external domains
* rtp payload was a bit different from usual VoIP traffic (video
parts, application sharing, file downloads etc.), but SDP was fine
according to RFC, and OpenSER mediaproxy worked also
* handling of peer-to-peer presence (SUBSCRIBE, MEASSAGE, OPTIONS)
* The number of messages to handle was not that much (some thousand
subs).
For my understanding this should also be possible with Freeswitch with
bypass_media. Right?
Best regards
Peter
Ognjen Seslija schrieb:
Quote: | Hello,
FS by design is B2BUA, and it cannot route INVITEs and other SIP
methods. It can however, bridge a-leg to b-leg with or w/o media and
doing plenty other cool stuff much better than commercial projects. I
suggest joining us on irc to detail your setup so we can help you.
Regards,
Ognjen (sekil on #freeswitch).
On Fri, May 22, 2009 at 7:24 AM, Rajagopal, Sridhar (Sridhar)
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