rtpproxy -l [PUBLIC-IP] [PRIVATE-IP] -s udp:127.0.0.1:7722 -u rtpproxy rtpproxy -p /var/run/rtpproxy/rtpproxy.pid -d DBUG LOG_LOCAL0
and in your config, (my case) something like:
modparam("rtpproxy", "rtpproxy_sock", "udp:localhost:7722")
then, when calls are being established, you need to check where it's coming from and where it's going, and execute rtproxy with the parameter of what IP to use, something like:
public ip:
rtpproxy_manage("cor", "[PUBLIC-IP]");
private ip:
rtpproxy_manage("co", "[PRIVATE-IP]");
good luck.
Hello,
I mean Internal facing to Asterisk(s)
External - facing to Provider(s)
On 10/01/2019 12:11, YASIN CANER wrote:
> Hello,
>
> Kamailio support 2 backend for rtp. RTP Engine and RTP proxy. it depends
> your configuration. I read from your post that
> "I am not sure if this is an _RTP engine _issue and how to resolve
> this." . I thought it is rtpengine.
>
> Indeed , they are different services and has different configuration
> that do same thing that relays rtp packets. you can find examples from
> documentation.
>
> I didnt get internal interface what is mean? but if rtpproxy/engine
> relays to Asterisk on -lo interface , you should listen your -lo
> interface by Wireshark.
>
> Good luck.
>
> Yasin CANER
>
>
> ------------------------------------------------------------------------
> *From:* sr-users <sr-users-bounces@lists.kamailio.org> on behalf of
> Serge S.Yuriev <me@nevian.org>
> *Sent:* Thursday, January 10, 2019 11:30 AM
> *To:* Kamailio (SER) - Users Mailing List
> *Subject:* Re: [SR-Users] No Media in SIP Incoming calls
> Hi
>
> - I see you mentioned rtpengine but config shows rtpproxy - which one
> you use? Maybe you mixed things?
> - Have you tried to capture internal interface of kamailio machine? Is
> RTP there?
>
> --
> Wbr, Serge via mobile
>
> 09.01.2019, 10:37, "Prashant Gupta" <prashant@farmguide.in>:
>> Hi,
>> I have the following architecture - SIP provider <-> Kamailio <->
>> Asterisk servers
>> Currently I have everything setup and incoming calls from Sip are
>> routed to my asterisk server. The issue is however that when I answer
>> the call, there is no media in the call. I have tried connecting with
>> a normal local extension(not SIP,eg 1001) and there is a normal flow
>> of media.
>> When i try to sniff my connection via Wireshark on the asterisk
>> server, there is an outflow of RTP packets but the same RTP traffic
>> does not appear on the Wireshark of my Kamailio server connection.
>> I am not sure if this is an RTP engine issue and how to resolve this.
>> I have -
>> modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:45038
>> <http://127.0.0.1:45038/>")
>> this in my kamailio cfg but I don;t know which port to use here.
>> Any suggestions?
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users@lists.kamailio.org <mailto:sr-users@lists.kamailio.org>
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
--
Serge S. Yuriev
Senior VoIP engineer
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users@lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users