Re: [SR-Users] question about rtpproxy "r" flag

2013-07-12 Thread Alex Hermann 
On Friday 12 July 2013 06:59:38 Juha Heinanen wrote:
> rtpproxy_offer and answer functions have "r" flag described as follows:
> 
>   r - flags that IP address in SDP should be trusted. Without this flag,
>   rtpproxy ignores address in the SDP and uses source address of the SIP
>   message as media address which is passed to the RTP proxy.
> 
> how does rtpproxy "use" this ip address?

It directly starts forwarding packets to it. If you don't use this flag, 
rtpproxsy will wait until it has received poackets from both parties in the 
call.


> if sip message comes from behind nat, ip address in sdp is local
> address, not the address where rtp packets come from.  and if this
> request has passed another (e.g. outbound) proxy before hitting the
> current one, also source address of sip message is not the address where
> rtp packets come from.  either address thus seems to be useless for
> rtpproxy.

You can make the addres more "usefull" by rewriting the address with 
fix_natted_sdp() on the first proxy the UAC reaches (load-balancer). If you 
don't have that possibility, maybe you can call fix_natted_sdp() on the proxy 
itself and call msg_apply_changes() before invoking the rtpproxy.
-- 
Alex Hermann

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Re: [SR-Users] question about rtpproxy "r" flag

2013-07-12 Thread Juha Heinanen 
Alex Hermann writes:

> > rtpproxy_offer and answer functions have "r" flag described as follows:
> > 
> >   r - flags that IP address in SDP should be trusted. Without this flag,
> >   rtpproxy ignores address in the SDP and uses source address of the SIP
> >   message as media address which is passed to the RTP proxy.
> > 
> > how does rtpproxy "use" this ip address?
> 
> It directly starts forwarding packets to it. If you don't use this flag, 
> rtpproxsy will wait until it has received poackets from both parties in the 
> call.

it is not what the above description tells.  it just tells where
rtpproxy takes the address, nothing about when it starts to send
packets.  proper use of rtpproxy or mediaproxy-ng is difficult until
this has been clarified.  waiting for both parties to send something
first, would not work if the other party is muted.

> > if sip message comes from behind nat, ip address in sdp is local
> > address, not the address where rtp packets come from.  and if this
> > request has passed another (e.g. outbound) proxy before hitting the
> > current one, also source address of sip message is not the address where
> > rtp packets come from.  either address thus seems to be useless for
> > rtpproxy.
> 
> You can make the addres more "usefull" by rewriting the address with 
> fix_natted_sdp() on the first proxy the UAC reaches (load-balancer). If you 
> don't have that possibility, maybe you can call fix_natted_sdp() on the proxy 
> itself and call msg_apply_changes() before invoking the rtpproxy.

it would result in double rewrite of the sdp when one motivation of
using mediaproxy-ng is that it does the rewriting only once.  there
should be possibility pass sdp ip address as parameter in offer and
answer calls.

-- juha

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Re: [SR-Users] question about rtpproxy "r" flag

2013-07-12 Thread Richard Fuchs 
On 07/12/13 05:48, Juha Heinanen wrote:

> it is not what the above description tells.  it just tells where
> rtpproxy takes the address, nothing about when it starts to send
> packets.  proper use of rtpproxy or mediaproxy-ng is difficult until
> this has been clarified.  waiting for both parties to send something
> first, would not work if the other party is muted.

In case of mediaproxy-ng, it doesn't wait. If it hasn't received
anything from the other endpoint yet, then it will use whatever address
it was given (either from the SDP or the SIP source address). Once it
has received a packet from the other endpoint, it will use this source
address. That's about as good as it gets without ICE.

> it would result in double rewrite of the sdp when one motivation of
> using mediaproxy-ng is that it does the rewriting only once.  there
> should be possibility pass sdp ip address as parameter in offer and
> answer calls.

This is possible through the second parameter to
rtpproxy_offer/answer/manage(). Mediaproxy-ng doesn't support this yet
(we have no use for it, auto-discovery works well enough), but it's next
up on my todo list for compatibility's sake.

cheers



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Re: [SR-Users] question about rtpproxy "r" flag

2013-07-12 Thread Juha Heinanen 
Richard Fuchs writes:

> In case of mediaproxy-ng, it doesn't wait. If it hasn't received
> anything from the other endpoint yet, then it will use whatever address
> it was given (either from the SDP or the SIP source address). Once it
> has received a packet from the other endpoint, it will use this source
> address. That's about as good as it gets without ICE.

thanks for the explanation.

> > it would result in double rewrite of the sdp when one motivation of
> > using mediaproxy-ng is that it does the rewriting only once.  there
> > should be possibility pass sdp ip address as parameter in offer and
> > answer calls.
> 
> This is possible through the second parameter to
> rtpproxy_offer/answer/manage(). Mediaproxy-ng doesn't support this yet
> (we have no use for it, auto-discovery works well enough), but it's next
> up on my todo list for compatibility's sake.

ok,

-- juha

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Re: [SR-Users] Carrier Failoverand LCR

2013-07-12 Thread Keith Hubner 
Hi,

Just a quick question, will the LCR module provide sufficient failover 
capacity? I have used the dispatcher module and like the way it marks a peer as 
down if it can't be contacted however I need to utilise the LCR functionality.

Maybe a combination of the both will be best?

Thanks,
Keith
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Re: [SR-Users] Carrier Failoverand LCR

2013-07-12 Thread Juha Heinanen 
Keith Hubner writes:

> Just a quick question, will the LCR module provide sufficient failover
> capacity? I have used the dispatcher module and like the way it marks
> a peer as down if it can't be contacted however I need to utilise the
> LCR functionality.

you can mark an unresponsive peer down also with lcr module by making
defunct_gw(period) call from config script when peer does not respond.

-- juha

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Re: [SR-Users] Asterisk cluster with Kamailio for load balancing.

2013-07-12 Thread Stoyan Mihaylov 
We use Kamailio for registration of users and loadbalancing.
I think you can start with next steps:
1. Single installation of kamailio with registration of users, fixing NAT
problems etc.
2. After you get some experience with Kamailio - you can add Asterisk
servers.
In asterisk servers - I accept all calls which comes from Kamailio -
registration of users is purely Kamailio "job".
In Kamailio - again I accept all calls which come from any of Asterisk
servers.



On Thu, Jul 11, 2013 at 8:10 PM, Володимир Іванець  wrote:

> Thanks a lot.
>
> Regards, Volodya Ivanets.
>
>
> 2013/7/11 Charles Chance 
>
>> I'm not aware of anything, but others may be.
>>
>> However, before diving straight into the Asterisk/Kamailio integration,
>> may I suggest you familiarise yourself first with Kamailio on its own. In
>> my opinion, this will give you a far better understanding of how it all
>> fits together, and the learning curve will be much more manageable.
>>
>> Any specific questions you have as you go along can be asked here.
>>
>> Cheers,
>>
>> Charles
>>
>>
>>
>> On 11 July 2013 16:50, Володимир Іванець wrote:
>>
>>> Hello,
>>>
>>> Yes I have. But I was hoping find some tutorial with experience sharing.
>>> Because I'm new to Kamailio, official tutorial did not cower all my
>>> questions.
>>>
>>> Thanks.
>>>
>>> Regards, Volodya Ivanets.
>>>
>>>
>>> 2013/7/11 Charles Chance 
>>>
 Hello,

 Have you looked at
 http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb
 ?

 Charles


 On 11 July 2013 16:01, Володимир Іванець wrote:

> Hello everyone!
>
> Recently I became interested in concept of putting couple of Asterisks
> servers in cluster. So I tried to replicate this scenario in testing
> environment. I was using database back-end for Asterisks settings and
> statistics and Kamailio as a load balancer.
>
> Obviously it did not worked [image: :cry:] . There is possible to
> make so many mistakes. Now I'm trying to find some step by step tutorial,
> but nothing yet.
>
> If someone experienced this, please advice. And if someone have useful
> links, please share.
>
> Thanks a lot.
>
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>> office: Unit 10 iBIC, Birmingham Science Park, Holt Court South, Birmingham
>> B7 4EJ.
>>
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[SR-Users] tls client to udp proxy

2013-07-12 Thread hiro 
hi

I have set up tls on kamailio successfully, but when I relay a TLS
client to an other proxy ($ru = "sip:" + $rU + "@" +
"127.0.0.1:5070;transport=udp";) via udp I get error messages.

>From the timing it seems like this warning always appears after I get
200OK from proxy, the Invite gets sent correctly, but kamailio seems
to fail while sending back the ACK.

18(11231) WARNING:  [forward.c:270]: WARNING: get_send_socket:
protocol/port mismatch (forced udp:77.180.86.63:5060, to
tls:127.0.0.1:5070)

TCP->UDP works.
TCP->TCP works.
UDP->UDP works.
UDP->TCP works.
TLS->TCP works.
TLS->UDP doesn't work.

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Re: [SR-Users] question about rtpproxy "r" flag

2013-07-12 Thread Richard Fuchs 
On 07/12/13 08:32, Richard Fuchs wrote:
> On 07/12/13 05:48, Juha Heinanen wrote:

>> it would result in double rewrite of the sdp when one motivation of
>> using mediaproxy-ng is that it does the rewriting only once.  there
>> should be possibility pass sdp ip address as parameter in offer and
>> answer calls.
> 
> This is possible through the second parameter to
> rtpproxy_offer/answer/manage(). Mediaproxy-ng doesn't support this yet
> (we have no use for it, auto-discovery works well enough), but it's next
> up on my todo list for compatibility's sake.

This is now implemented in both rtpproxy-ng and mediaproxy-ng.

cheers



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Re: [SR-Users] Asterisk cluster with Kamailio for load balancing.

2013-07-12 Thread Vitaliy Aleksandrov 

Indeed, this is a very good suggestion.

I can propose a little plan for the first step:
1. Learn how sip registration works
2. Configure kamailio as a sip registrar server.
3. Try to route calls between registered phones.
4. Read about in-dialog requests routing (record_route(), loose_route()) 
and configure calls between connected endpoint in a "right" way.


And if you want to learn mode:
5. Try to handle NAT issues at kamailio (symmetric sip / rtp relay)

Asterisk supports symmetric rtp pretty well, so if you want to route all 
calls through asterisk boxes you can skip rtp relay configuration step.



We use Kamailio for registration of users and loadbalancing.
I think you can start with next steps:
1. Single installation of kamailio with registration of users, fixing 
NAT problems etc.
2. After you get some experience with Kamailio - you can add Asterisk 
servers.
In asterisk servers - I accept all calls which comes from Kamailio - 
registration of users is purely Kamailio "job".
In Kamailio - again I accept all calls which come from any of Asterisk 
servers.






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