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Editorial Team
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Asked: 2026-06-04T10:29:33+00:00

I have a network set up with 2 UACs behind a NAT (one is

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I have a network set up with 2 UACs behind a NAT (one is Jitsi and the other is my own UA coded in C with SIP support). Asterisk is set up on a public IP.

In sip.conf I have the following for both UACs: nat=yes directmedia=no

I then execute my code which calls the Jitsi client. I answer via Jitsi and everything seems great.
Asterisk provides each UAC an RTP host/port to send video to via an SDP packet which I’ve parsed appropriately (audio is not included in this session). Each user agent begins transmitting RTP packets.

Here’s where the problem occurs:
Asterisks starts printing out

“Got RTP packet from XX.XXX.XX.XXX:XXXXX (type XX, seq XXXXX, ts XXXXXx, len XXXXXX)”

repeatedly from both UACs but never actually sends any of the RTP packets anywhere (I would expect “Send RTP ….”).

I’ve tested my H264 RTP code through a QuickTime Broadcasting Server and the packets properly decode on the local network through various media players. My SIP call seems to have no problems connecting and Asterisk never prints out any Warnings or Errors in the console.

I cannot for the life of me figure out why Asterisk is not forwarding the RTP packets. Any help would be appreciated.

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1 Answer

  1. Editorial Team

    It seems I needed to use STUN in order to find my public IP address.

    1. I switched from Asterisk to FreeSWITCH. Though this did not solve my problem, I found FreeSWITCH much easier to configure and debug which helped in my search.

    2. Turns out that neither FreeSWITCH nor Asterisk really care what IP address you provide them during a SIP session, they’ll simply use the IP address/port that a packet is received on (via the IP header) for communication. This confused me because I could make a call between 2 UACs but after a call was answered the audio/video media would never arrive at the correct destination. Turns out RTP implementations will use the IP address provided in the SDP message when setting up a media stream, regardless of what IP address SIP was using for its communication.

    In the end, I used STUN to discover the public IP/port for the calling UAC and inserted this correct IP/port combo into the SDP message. Subsequently when the answering UAC (Jitsi) set up its media stream it now had the correct IP/port combo to send/receive media from.

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