[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=259924#comment-259924 ] 

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

We appreciate the difficulties you are facing, however information request type issues would be better served in a different forum.

The Asterisk community provides support over IRC, mailing lists, and forums as described at http://asterisk.org/community. The Asterisk issue tracker is used specifically to track issues concerning bugs and documentation errors.

If this issue is actually a bug please use the Bug issue type instead.

Please see the Asterisk Issue Guidelines [1] for instruction on the intended use of the Asterisk issue tracker.

Thanks!

[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines

> 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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