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

[Asterisk Team (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-30174?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=259925#comment-259925 ] 

Asterisk Team commented on ASTERISK-30174:
------------------------------------------

Thanks for creating a report! The issue has entered the triage process. That means the issue will wait in this status until a Bug Marshal has an opportunity to review the issue. Once the issue has been reviewed you will receive comments regarding the next steps towards resolution. Please note that log messages and other files should not be sent to the Sangoma Asterisk Team unless explicitly asked for. All files should be placed on this issue in a sanitized fashion as needed.

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> 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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