[SR-Users] [sr-dev] kamailio SIP and RTP proxy

Henning Westerholt hw at skalatan.de
Tue Dec 1 11:07:01 CET 2020

Hi Patrick,

better to contact our sr-users list with the usage related questions, added to CC.

Have a look to the SDP of the SIP packets to see if it contains the correct IP would be one idea to debug this further.

Feel free to ask again on sr-users after you have got more details.



Henning Westerholt – https://skalatan.de/blog/
Kamailio services – https://gilawa.com<https://gilawa.com/>

From: sr-dev <sr-dev-bounces at lists.kamailio.org> On Behalf Of Patrick Leybag
Sent: Wednesday, November 25, 2020 6:26 AM
To: sr-dev at lists.kamailio.org
Subject: [sr-dev] kamailio SIP and RTP proxy

Hi, Can someone help me?
I self host a kamailio using my raspberry pi as a load balancer for my two asterisk servers and get a did number. when I call to my DID number it points to my kamailio and kamailio will distribute to asterisk server but the call has no audio. I tried port forwarding ports 5060 for SIP and 10000-20000 for RTP but it still does not work.

Any help is much appreciated. Thank you in advance

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