[asterisk-bugs] [JIRA] (ASTERISK-30174) RTP stream does not open between peers

[Irtaza Waheed (JIRA)]

Irtaza Waheed created ASTERISK-30174:
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             Summary: RTP stream does not open between peers
                 Key: ASTERISK-30174
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-30174
             Project: Asterisk
          Issue Type: Information Request
      Security Level: None
          Components: . I did not set the category correctly.
    Affects Versions: 16.13.0
         Environment: Raspberry Pi 3B with Raspbian 10 Buster running Asterisk
            Reporter: Irtaza Waheed


Good day,

I am happy to join this great community and hope that this mature community will help me learn better.

I have two peers with PJSIP accounts connected to the Raspberry Pi. I can call from one account to the other. However, no communiction/voice transmission occurs between the two. The ASterisk CLI reads as follows:

-- PJSIP/24-0000000f Internal Gosub(func-apply-sipheaders,s,1) complete GOSUB_RETVAL=                                  
-- Called PJSIP/24/sip:24 at 192.168.200.210:5060;ob                                                                     == Using SIP RTP Audio TOS bits 184                                                                                     == Using SIP RTP Audio TOS bits 184 in TCLASS field.                                                                    == Using SIP RTP Audio CoS mark 5                                                                                         -- PJSIP/24-0000000f answered PJSIP/23-0000000e                                                                            > 0x7410d310 -- Strict RTP learning after remote address set to: 192.168.200.210:4006                                   
> 0x74113cd0 -- Strict RTP learning after remote address set to: 192.168.200.230:4006                               
-- Channel PJSIP/24-0000000f joined 'simple_bridge' basic-bridge <0e0ed803-5d80-4594-b8a0-4d330d89d0f6>                 
-- Channel PJSIP/23-0000000e joined 'simple_bridge' basic-bridge <0e0ed803-5d80-4594-b8a0-4d330d89d0f6>                    > 0x7410d310 
-- Strict RTP switching to RTP target address 192.168.200.210:4006 as source                               
> 0x7410d310 -- Strict RTP learning complete - Locking on source address 192.168.200.210:4006      

But no communication happens. The problem is that I don't see that any stream has been opened or selected. Like:

-- Strict RTP qualifying stream type: audio

I see this informartion when I call a peer from another PJSIP account and voice transmission occurs. 

Am I missing something?

Thank you in advance for any help. 






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