2 Way RTP is ON, but still one way audio issue

Asterisk Asterisk SIP
1

I have one way audio issue, but “rtp debug” says both way traffic are on but caller side is not being heard by the b-side. I am running asterisk 16.30.1 on CentOS 7; Can someone throw some light?

rtp debug is like this:

Got RTP packet from 1.2.3.4:27428 (type 00, seq 000768, ts 3431796719, len 000160)
Sent RTP packet to 5.6.7.8:37272 (type 08, seq 009156, ts 3431796712, len 000160)

Got RTP packet from 5.6.7.8:37272 (type 08, seq 051647, ts 1224747680, len 000160)
Sent RTP packet to 1.2.3.4:27428 (type 00, seq 001942, ts 1224747680, len 000160)

2

Check your firewall, disable SIP Alg if on

3

SIP ALG is disabled on the router. any other parameter on the asterisk config files that might need a tweak ? Any suggestions.

4

in /etc/asterisk/rtp.conf
change to:
[general]
rtpstart=10000
rtpend=25000
icesupport=true
stunaddr:19302=stun.l.google.com

5

Please enable rtp debug, Then make call inbound or outbound calls. After that it will show you Signal voice IPs. Then add that IPs as a static route. It will work I am sure. Restart network service and check it.

6

Thanks guys, I made changes to my rtp.conf as suggested and i also added static routes to all the signaling IPs showing up on the rtp debug. Whichever fixed it, the issue is solved. Appreciate your lead guys. Thank you.

[general]
rtpstart=10000
rtpend=25000
icesupport=true
stunaddr:19302=stun.l.google.com

7

Good news.

8

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