[SR-Users] One way audio on calls from pstn

[dotnetdub]

On 1 July 2010 23:03, Dmitri Korotkov <dmitri.korotkov at festart.ee> wrote:

>  Hi,
>
> default kamailio config file(its routing part) already has rtpproxy support
> in case if WITH_NAT is defined.
> And there is no problems when NATed subscribers calls one to other...
> I have problem only with PSTN and only with incoming call.
>
> BR,
> Dmitri
>
>


I don't think rtpproxy is being engaged in your inbound route.




>  02.07.2010 0:51, dotnetdub пишет:
>
>
>
> On 1 July 2010 22:41, Dmitri Korotkov <dmitri.korotkov at festart.ee> wrote:
>
>> Hi,
>>
>> voice:/# ps auxf |grep rtpproxy |grep -v grep
>> rtpproxy  1291  0.0  0.0  26800   876 ?        Ssl  Jun18   0:10
>> /usr/sbin/rtpproxy -u rtpproxy rtpproxy -l my.public.ip.here -s
>> udp:localhost 7722
>> voice:/#
>>
>>
>> kamailio.cfg:
>> #!define WITH_MYSQL
>> #!define WITH_AUTH
>> #!define WITH_ACCDB
>> #!define WITH_NAT
>> #!define WITH_PSTN
>>
>> #!ifdef WITH_NAT
>> loadmodule "nathelper.so"
>> #!endif
>>
>> # ----- nathelper -----
>> #!ifdef WITH_NAT
>> modparam("nathelper", "rtpproxy_sock", "udp:127.0.0.1:7722")
>> modparam("nathelper", "natping_interval", 30)
>> modparam("nathelper", "ping_nated_only", 1)
>> modparam("nathelper", "sipping_bflag", 7)
>> modparam("nathelper", "sipping_from", "sip:pinger at kamailio.org"<sip:pinger at kamailio.org>
>> )
>> modparam("registrar|nathelper", "received_avp", "$avp(i:80)")
>> modparam("usrloc", "nat_bflag", 6)
>> #!endif
>>
>>
>>
>> 02.07.2010 0:32, dotnetdub пишет:
>>
>>
>>
>>    I'm not overly familiar with rtpproxy as we use mediaproxy but you
> will need to engage it somewhere in your script, are you doing that?
>
>  Take a look at
> http://www.voip-info.org/wiki/view/Kamailio+1.5.x+and+RTPProxy
>
>  Can you see any rtpproxy messages in syslog?
>
>
>
>
>>   On 1 July 2010 21:53, Dmitri Korotkov <dmitri.korotkov at festart.ee>wrote:
>>
>>> Hello,
>>>
>>> I have kamailio installation WITH_PSTN, WITH_NAT and rtpproxy.
>>> Using following scenario:
>>> [kamailio]<-sip trunk ->[asterisk gw] <->sip trunk <-> [PSTN provider]
>>>
>>> All kamailio sip subscribers are behind nat in different networks.
>>>
>>> 1. OK. Local kamailio users can call one to other even they are on
>>> different networks behind nat.
>>> 2. OK. Outgoing calls from kamailio users to PSTN work also very well.
>>> 3. Not OK.  Incoming from PSTN side calls have only one way audio.
>>>
>>> I tcpdump'ed kamailio box and found, that pstn provider sends RTP packets
>>> to kamailio IP in case of answered call.
>>>
>>> I guess that rtpproxy is not active in case of pstn call.  Is it true ?
>>>
>>> I am using more less default kamailio config
>>>
>>> Could you please suggest solution ?
>>>
>>> BR,
>>> Dmitri
>>>
>>>
>>
>>  Hi Dmitri,
>>
>>  Check out the nathelper module.
>>
>>  Regards,
>> Brian
>>
>>
>>
>
>
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