[SR-Users] No Media in SIP Incoming calls

[Serge S. Yuriev]

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 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?
>>
>> _______________________________________________
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>>

-- 
Serge S. Yuriev
Senior VoIP engineer