[asterisk-bugs] [JIRA] (ASTERISK-30174) RTP stream does not open between peers
Asterisk Team (JIRA)
noreply at issues.asterisk.org
Wed Aug 10 10:45:09 CDT 2022
[ https://issues.asterisk.org/jira/browse/ASTERISK-30174?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Asterisk Team closed ASTERISK-30174.
------------------------------------
Resolution: Not A Bug
> 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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