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Summary:ASTERISK-30174: RTP stream does not open between peers
Reporter:Irtaza Waheed (iwaheed)Labels:
Date Opened:2022-08-10 10:44:06Date Closed:2022-08-10 10:44:09
Priority:MinorRegression?
Status:Closed/CompleteComponents:. I did not set the category correctly.
Versions:16.13.0 Frequency of
Occurrence
Related
Issues:
Environment:Raspberry Pi 3B with Raspbian 10 Buster running AsteriskAttachments:
Description: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@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.


Comments:By: Asterisk Team (asteriskteam) 2022-08-10 10:44:08.844-0500

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
By: Asterisk Team (asteriskteam) 2022-08-10 10:44:12.987-0500

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.

A good first step is for you to review the [Asterisk Issue Guidelines|https://wiki.asterisk.org/wiki/display/AST/Asterisk+Issue+Guidelines] if you haven't already. The guidelines detail what is expected from an Asterisk issue report.

Then, if you are submitting a patch, please review the [Patch Contribution Process|https://wiki.asterisk.org/wiki/display/AST/Patch+Contribution+Process].

Please note that once your issue enters an open state it has been accepted. As Asterisk is an open source project there is no guarantee or timeframe on when your issue will be looked into. If you need expedient resolution you will need to find and pay a suitable developer. Asking for an update on your issue will not yield any progress on it and will not result in a response. All updates are posted to the issue when they occur.

Please note that by submitting data, code, or documentation to Sangoma through JIRA, you accept the Terms of Use present at [https://www.asterisk.org/terms-of-use/|https://www.asterisk.org/terms-of-use/].