No RTP traffic, ports enabled, no sound

Asterisk Asterisk SIP
1

Hello, I am having issue with both external and internal calls not having any audio, most likely due to me not receiving any RTP traffic (I checked using rtp set debug on ). I am using ufw so naturally I have enabled ports in range 10000-20000/udp as can be seen on this output:

root@mainasteriskenv:~# sudo ufw status 
Status: active

To                         Action      From
--                         ------      ----
5060/tcp                   ALLOW       Anywhere                  
22/tcp                     ALLOW       Anywhere                  
5060                       ALLOW       Anywhere                  
5061                       ALLOW       Anywhere                  
5061/udp                   ALLOW       Anywhere                  
5060/udp                   ALLOW       Anywhere                  
4000:5000/udp              ALLOW       Anywhere                  
4000:5000/tcp              ALLOW       Anywhere                  
10000:20000/udp            ALLOW       Anywhere                  
53/udp                     ALLOW       Anywhere                  
53/tcp                     ALLOW       Anywhere                  
123/udp                    ALLOW       Anywhere                  
5060:5061/udp              ALLOW       Anywhere                  
5060/tcp (v6)              ALLOW       Anywhere (v6)             
22/tcp (v6)                ALLOW       Anywhere (v6)             
5060 (v6)                  ALLOW       Anywhere (v6)             
5061 (v6)                  ALLOW       Anywhere (v6)             
5061/udp (v6)              ALLOW       Anywhere (v6)             
5060/udp (v6)              ALLOW       Anywhere (v6)             
4000:5000/udp (v6)         ALLOW       Anywhere (v6)             
4000:5000/tcp (v6)         ALLOW       Anywhere (v6)             
10000:20000/udp (v6)       ALLOW       Anywhere (v6)             
53/udp (v6)                ALLOW       Anywhere (v6)             
53/tcp (v6)                ALLOW       Anywhere (v6)             
123/udp (v6)               ALLOW       Anywhere (v6)             
5060:5061/udp (v6)         ALLOW       Anywhere (v6)             

root@mainasteriskenv:~#

this range corresponds to rtp.conf

Note that I am using fail2ban as well, though that should have zero impact on anything. But I feel like I should mention it…

Thank you to anyone who contributes here

2

An additional reason is that the IP address in the SDP is incorrect, and so the remote side sends the RTP to the wrong place. You should check the output of the respective SIP logging (“pjsip set logger on” in PJSIP, “sip set debug on” in chan_sip) and confirm it.

3

I don’t see any SDP argument in “sip set debug on”, I am using chansip so that’s weird.

only SDP related stuff I see is:

adding codec ulaw to SDP
and
application/sdp though in the contents I don’t see anything that would ring any bells

4

The SDP is contained within the SIP messages themselves.

5

In the whole log, that is, from Dial() to Hangup() only 2 IP addresses appear, that is my server IP and my soft phone IP, so that should be in order. Both extensions have correct IPs assigned to them ( my softphone is ext. 2001, server being 3002), no mismatch there. I am happy to send a dump of the debug here, though I’m not sure if it’s worth doing it here in public

6

bump

7

is there anything else I could do or provide here? I’m starting to get really desperate.

8

I don’t have anything else to ask for or add to this thread as of this time.

9

alright, thank you.

10

Hypothetically if this was to be the issue, what would be the steps to find the core issue and fix it?

11

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