[asterisk-bugs] [JIRA] (ASTERISK-29105) chan_pjsip: 180 Ringing with SDP not changed into progress

[Holger Hans Peter Freyther (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29105?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=253312#comment-253312 ] 

Holger Hans Peter Freyther commented on ASTERISK-29105:
-------------------------------------------------------

The chan_pjsip response handler will be called twice and we queue the AST_CONTROL_PROGRESS twice. In contrast to the 180 we generate progress twice (two 183 on the wire).

I worked around this by queuing the progress only once and only for the AST_SIP_SESSION_AFTER_MEDIA case. Is there a better way of achieving this?

The change is in gerrit. https://gerrit.asterisk.org/c/asterisk/+/15307

> chan_pjsip: 180 Ringing with SDP not changed into progress
> ----------------------------------------------------------
>
>                 Key: ASTERISK-29105
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29105
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_sdp_rtp
>    Affects Versions: 17.6.0
>         Environment: Debian 10
>            Reporter: Sebastian Damm
>            Severity: Minor
>         Attachments: asterisk180SDP.tgz, asterisk.log
>
>
> When Asterisk receives an 180 Ringing response with SDP, it does not forward this response but instead sends an 180 Ringing response without SDP to the caller. Additionally, it does not forward the audio from B to the caller.
> I would expect Asterisk to forward the Ringing with SDP as well as the audio from B to A.
> Attached is a log of Asterisk showing the behavior. Additionally, I have attached a docker scenario to reproduce it. Follow these steps to reproduce:
> * docker-compose up -d
> * docker-compose exec sipp /bin/bash
> * /testcase/start.sh
> * exit from container
> * docker-compose logs asterisk
> * Inside the sipp container you will find sipp output from both caller and called, as well as a pcap file of all udp traffic. Inside the pcap file, you can see that B sends audio to the Asterisk, but there is no audio from Asterisk to A before the 200 OK arrives.



--
This message was sent by Atlassian JIRA
(v6.2#6252)