[Kamailio-Users] Ratelimit outgoing SIP INVITE messages after dispatcher

[Ovidiu Sas]

rl_check()/rl_check_pipe() will work on an *incoming* SIP packet, but
that SIP packet will go *out*, so what's the difference?  The
ratelimiting mechanism is flexible and it is entirely up to you how
you want to use it.
You can define several pipes and associate each pipe to a destination.
 Once you know where you want to route the call, just ratelimit the
traffic based on you predefined pipes.

Regards,
Ovidiu Sas

On Tue, Aug 19, 2008 at 12:09 PM, sergejf <serge at elasticall.com> wrote:
>
> From what I understand rl_check_pipe() will work on all *incoming* INVITE
> packets (assuming we use that setup) while what we want is limiting the rate
> on *outgoing* SIP trunks that have different CPS ratings. For example,
> carrier 1 supports 10 CPS per trunk while carrier 2 only accepts 5 CPS.
>
> For argument's sake let's assume we can get by with evenly spreading the CPS
> load between SIP trunks with identical CPS ratings (e.g. 10 SIP trunks at 10
> CPS for an aggregate 100 CPS). The only possibility I can see is calling
> rl_check_pipe() after havinig initialized ratelimit as follows
>
> modparam("ratelimit", "timer_interval", 5)
> modparam("ratelimit", "queue", "1:INVITE")
> modparam("ratelimit", "queue", "2:*")
> modparam("ratelimit", "pipe", "1:RED:100")
>
> then using carrierroute with 10% load factors on every SIP trunk
> (destination gateway). This should evenly spread a 100 CPS load between 10
> SIP trunks rated at 10 CPS each, correct?
>
> Nevertheless this does not support a configuration with SIP trunks that have
> all different CPS ratings.
>
>
> Ovidiu Sas-3 wrote:
>>
>> Just call rl_check()/rl_check_pipe() to see if the imposed cps limit
>> was reached:
>>  - if not, route the call normally;
>>  - if yes, pick the next destination/carrier.
>>
>> Regards,
>> Ovidiu Sas
>>
>> On Sat, Aug 16, 2008 at 5:48 PM, sergejf <serge at elasticall.com> wrote:
>>>
>>> Hello,
>>>
>>> We are connected to several VoIP carriers that impose a calls per second
>>> (CPS) rate limit on their SIP trunks. For example, on TRUNK1 we can only
>>> make call attempts at 10 CPS.
>>>
>>> Our calls are originally made by a dialer that does not support rate
>>> limiting, hence it always blasts calls as fast as it can. Upon receiving
>>> these SIP INVITE messages we determine what carrier to use and perform a
>>> mapping to a dispatcher group. Call attempts are then load-balanced
>>> between
>>> all SIP trunks from the chosen carrier.
>>>
>>> My question is whether it is possible to use the ratelimit module in
>>> conjunction with the dispatcher (or carrierroute) to ensure we don't
>>> overload the carrier SIP trunks? If this is possible, then I wonder how
>>> to
>>> configure ratelimit to support a 10 CPS call attempts rate.
>>>
>>> Thanks in advance for any advice on the topic!
>>>
>>> Serge
>>
>>
>
> --
> View this message in context: http://www.nabble.com/Ratelimit-outgoing-SIP-INVITE-messages-after-dispatcher-tp19000383p19053566.html
> Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
>
>
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