[SR-Users] No Media in SIP Incoming calls
david.villasmil.work at gmail.com
Thu Jan 10 15:10:04 CET 2019
As previously stated, you probably are not activating you rtpproxy. When
kamailio starts up, it checks whether it can connect to the local rtpproxy.
Check your kamailio log for errors.
You also need to have rtpproxy running on your kamailio box, on the
specified port (or setting the correct port on the config file).
If your asterisk is on a private network, you will need to start rtpproxy
specifying the outside and the inside (private) rtp ips, like:
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:
email: david.villasmil.work at gmail.com
On Thu, Jan 10, 2019 at 12:33 PM Serge S. Yuriev <me at nevian.org> wrote:
> 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 at lists.kamailio.org> on behalf of
> > Serge S.Yuriev <me at 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 at 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 at lists.kamailio.org <mailto:sr-users at 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 at lists.kamailio.org
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