19 Aug
2008
19 Aug
'08
9:07 p.m.
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(a)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(a)elasticall.com> wrote:
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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Our calls are originally made by a dialer that does not support rate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