14 May
2007
14 May
'07
9:58 p.m.
I may be misunderstanding things (very probably so), but all the
examples for both SER and OpenSER that I have seen either do not do
NAT or NAT but are not the "end point" for a UAC to the PSTN. From
what I can tell they are just sending on the SIP request to the next
destination as is. The setup I am trying to accomplish is
* SIP clients are all on a private network with connectivity to
OpenSER directly on one interface (on the same private network)
* OpenSER's other interface is on the external network (internet
facing)
* SIP clients are only sending telephone numbers
(sip:telephone_number@*... do not know about PSTN providers)
* OpenSER connects to the PSTN provider and send the number to
dial that came from the SIP client
* OpenSER "proxies" the entire call (with the help of rtpproxy or
mediaproxy for RTP of course)
* No incoming calls from the internet to OpenSER (no support for
that is needed)
* Registration not required for sip clients (they are all on same
private network and authorized)
I have found several posts, example configs, documents that have
pieces of what I need (from what I can tell).. and I have tried to
put it together, but it does not quite work...
So is there some example that fits this type of usage? If not one
then possibly several pieces from a few documents? I have been
thinking that OpenSER setup as a outbound proxy configured for
multihome, but everything I have seen on that just routes the calls
through to where ever the SIP client was requesting as a final
destination and I need to send to one destination (the PSTN
provider). Any help is greatly appreciated.
BTW, I was thinking the NAThelper or outbound proxy example from
http://www.voip-info.org/wiki-SER+tips+and+tricks
Looked close to what is needed, but have a bunch of stuff (seemingly)
unneeded for my scenario and have nothing about PSTN connectivity.
Thanks for taking the time to read this long post (if you made it
this far).
Taylor
14 May
14 May
11:06 p.m.
Hi Taylor,
From what you describe here, this should be an easy
setup:
- configure all your SIP clients to talk to openser (on the private
interface)
- in the openser config, as soon as you get an INVITE, route the
INVITE to the appropriate PSTN GW (based on your rules). You can
hardcode them or use lcr.
- use rtpproxy to bridge the media
Hope this helps,
Ovidiu Sas
On 5/14/07, Taylor Carpenter <taylor(a)codecafe.com> wrote:
I may be misunderstanding things (very probably so),
but all the
examples for both SER and OpenSER that I have seen either do not do
NAT or NAT but are not the "end point" for a UAC to the PSTN. From
what I can tell they are just sending on the SIP request to the next
destination as is. The setup I am trying to accomplish is
* SIP clients are all on a private network with connectivity to
OpenSER directly on one interface (on the same private network)
* OpenSER's other interface is on the external network (internet
facing)
* SIP clients are only sending telephone numbers
(sip:telephone_number@*... do not know about PSTN providers)
* OpenSER connects to the PSTN provider and send the number to
dial that came from the SIP client
* OpenSER "proxies" the entire call (with the help of rtpproxy or
mediaproxy for RTP of course)
* No incoming calls from the internet to OpenSER (no support for
that is needed)
* Registration not required for sip clients (they are all on same
private network and authorized)
I have found several posts, example configs, documents that have
pieces of what I need (from what I can tell).. and I have tried to
put it together, but it does not quite work...
So is there some example that fits this type of usage? If not one
then possibly several pieces from a few documents? I have been
thinking that OpenSER setup as a outbound proxy configured for
multihome, but everything I have seen on that just routes the calls
through to where ever the SIP client was requesting as a final
destination and I need to send to one destination (the PSTN
provider). Any help is greatly appreciated.
BTW, I was thinking the NAThelper or outbound proxy example from
http://www.voip-info.org/wiki-SER+tips+and+tricks
Looked close to what is needed, but have a bunch of stuff (seemingly)
unneeded for my scenario and have nothing about PSTN connectivity.
Thanks for taking the time to read this long post (if you made it
this far).
Taylor
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15 May
15 May
12:40 a.m.
Thanks for your reply Ovidiu.
Is there any need to change any thing with the SIP request because of
NAT? Do I need to use rewritehost? Before the NAT requirements I was
doing ds_select_domain() for loadbalancing... At some point I will want
to do that again, but for the moment I am just wanting private lan
traffic to be able to place outbound calls with the PSTN provider.
On that note some (though not all) of the providers require auth and I
would like OpenSER to provide the credentials so that the SIP clients
only have to know about OpenSER to dial out... I have been using the uac
module for that.
So is there anything special (besides RTP traffic) to do with the rest
of the SIP packets before doing a t_relay()?
The examples I have seen using rtpproxy used FAEE, FAEI, FAIE, FAII,
etc.. I assume that all of those are not needed if everything is coming from
the private side out to the internet (and PSTN provider). Is that
correct?
Taylor
On Mon, May 14, 2007 at 04:06:50PM -0400, Ovidiu Sas wrote:
Hi Taylor,
From what you describe here, this should be an easy setup:
- configure all your SIP clients to talk to openser (on the private
interface)
- in the openser config, as soon as you get an INVITE, route the
INVITE to the appropriate PSTN GW (based on your rules). You can
hardcode them or use lcr.
- use rtpproxy to bridge the media
Hope this helps,
Ovidiu Sas
On 5/14/07, Taylor Carpenter <taylor(a)codecafe.com> wrote:
>I may be misunderstanding things (very probably so), but all the
>examples for both SER and OpenSER that I have seen either do not do
>NAT or NAT but are not the "end point" for a UAC to the PSTN. From
>what I can tell they are just sending on the SIP request to the next
>destination as is. The setup I am trying to accomplish is
>
> * SIP clients are all on a private network with connectivity to
>OpenSER directly on one interface (on the same private network)
> * OpenSER's other interface is on the external network (internet
>facing)
> * SIP clients are only sending telephone numbers
>(sip:telephone_number@*... do not know about PSTN providers)
> * OpenSER connects to the PSTN provider and send the number to
>dial that came from the SIP client
> * OpenSER "proxies" the entire call (with the help of rtpproxy or
>mediaproxy for RTP of course)
> * No incoming calls from the internet to OpenSER (no support for
>that is needed)
> * Registration not required for sip clients (they are all on same
>private network and authorized)
>
>I have found several posts, example configs, documents that have
>pieces of what I need (from what I can tell).. and I have tried to
>put it together, but it does not quite work...
>
>So is there some example that fits this type of usage? If not one
>then possibly several pieces from a few documents? I have been
>thinking that OpenSER setup as a outbound proxy configured for
>multihome, but everything I have seen on that just routes the calls
>through to where ever the SIP client was requesting as a final
>destination and I need to send to one destination (the PSTN
>provider). Any help is greatly appreciated.
>
>BTW, I was thinking the NAThelper or outbound proxy example from
>
> http://www.voip-info.org/wiki-SER+tips+and+tricks
>
>Looked close to what is needed, but have a bunch of stuff (seemingly)
>unneeded for my scenario and have nothing about PSTN connectivity.
>
>Thanks for taking the time to read this long post (if you made it
>this far).
>
>Taylor
12:55 a.m.
Hi Taylor,
see inline:
On 5/14/07, Taylor Carpenter <taylor(a)codecafe.com> wrote:
Thanks for your reply Ovidiu.
Is there any need to change any thing with the SIP request because of
NAT? Do I need to use rewritehost? Before the NAT requirements I was
doing ds_select_domain() for loadbalancing... At some point I will want
to do that again, but for the moment I am just wanting private lan
traffic to be able to place outbound calls with the PSTN provider.
Yes, you can use rewritehost() to route your request.
lcr would work better here.
On that note some (though not all) of the providers
require auth and I
would like OpenSER to provide the credentials so that the SIP clients
only have to know about OpenSER to dial out... I have been using the uac
module for that.
You can use the uac for authentication, but there are some
restrictions (your provider must accept authenticated INVITEs with the
same CSeq number).
So is there anything special (besides RTP traffic) to
do with the rest
of the SIP packets before doing a t_relay()?
The examples I have seen using rtpproxy used FAEE, FAEI, FAIE, FAII,
etc.. I assume that all of those are not needed if everything is coming from
the private side out to the internet (and PSTN provider). Is that
correct?
You will need to proxy the RTP traffic between the two interfaces.
Taylor
On Mon, May 14, 2007 at 04:06:50PM -0400, Ovidiu Sas wrote:
Hi Taylor,
From what you describe here, this should be an easy setup:
- configure all your SIP clients to talk to openser (on the private
interface)
- in the openser config, as soon as you get an INVITE, route the
INVITE to the appropriate PSTN GW (based on your rules). You can
hardcode them or use lcr.
- use rtpproxy to bridge the media
Hope this helps,
Ovidiu Sas
On 5/14/07, Taylor Carpenter <taylor(a)codecafe.com> wrote:
>I may be misunderstanding things (very probably so), but all the
>examples for both SER and OpenSER that I have seen either do not do
>NAT or NAT but are not the "end point" for a UAC to the PSTN. From
>what I can tell they are just sending on the SIP request to the next
>destination as is. The setup I am trying to accomplish is
>
> * SIP clients are all on a private network with connectivity to
>OpenSER directly on one interface (on the same private network)
> * OpenSER's other interface is on the external network (internet
>facing)
> * SIP clients are only sending telephone numbers
>(sip:telephone_number@*... do not know about PSTN providers)
> * OpenSER connects to the PSTN provider and send the number to
>dial that came from the SIP client
> * OpenSER "proxies" the entire call (with the help of rtpproxy or
>mediaproxy for RTP of course)
> * No incoming calls from the internet to OpenSER (no support for
>that is needed)
> * Registration not required for sip clients (they are all on same
>private network and authorized)
>
>I have found several posts, example configs, documents that have
>pieces of what I need (from what I can tell).. and I have tried to
>put it together, but it does not quite work...
>
>So is there some example that fits this type of usage? If not one
>then possibly several pieces from a few documents? I have been
>thinking that OpenSER setup as a outbound proxy configured for
>multihome, but everything I have seen on that just routes the calls
>through to where ever the SIP client was requesting as a final
>destination and I need to send to one destination (the PSTN
>provider). Any help is greatly appreciated.
>
>BTW, I was thinking the NAThelper or outbound proxy example from
>
> http://www.voip-info.org/wiki-SER+tips+and+tricks
>
>Looked close to what is needed, but have a bunch of stuff (seemingly)
>unneeded for my scenario and have nothing about PSTN connectivity.
>
>Thanks for taking the time to read this long post (if you made it
>this far).
>
>Taylor
11:04 p.m.
New subject: [Users] Re: openser setup as a telephony gateway (to PSTN) for NAT'd clients
After a good deal of experimenting I got calls going out and had 1-way
audio.. I was then able made changes and the audio would swap
directions in which it worked (callee or caller could talk). I made some
other adjustments but the main piece that made the difference and gave
me 2-way audio was taking rtpproxy out of bridge mode. Now everything
seems to be good. Thanks for the help.
Taylor
16 May
16 May
3:53 a.m.
New subject: [Users] Assigning multi-valued AVP to $ru
Openser 1.2 question:
Having a multi-valued AVP $avp(s:uri), does
$ru = $avp(s:uri);
achieve the same as
avp_pushto("$ru", "$avp(s:uri)/g"); ?
Or is there something like
$ru = $avp(s:uri)/g; ?
thanks,
Christian
11:07 a.m.
New subject: [Users] Assigning multi-valued AVP to $ru
Hello,
On 05/16/07 03:53, Christian Schlatter wrote:
Openser 1.2 question:
Having a multi-valued AVP $avp(s:uri), does
$ru = $avp(s:uri);
achieve the same as
avp_pushto("$ru", "$avp(s:uri)/g"); ?
Or is there something like
$ru = $avp(s:uri)/g; ?
no, for such case you still have to use
avp_pushto("$ru",
"$avp(s:uri)/g"). It is the reason why avp_pushto() stilll exists. On
the todo list is kind of "foreach" addition to overcome this.
Cheers,
Daniel
thanks,
Christian
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6 comments
4 participants
participants (4)
-
Christian Schlatter -
Daniel-Constantin Mierla -
Ovidiu Sas -
Taylor Carpenter