[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=252407#comment-252407 ] 

Holger Hans Peter Freyther commented on ASTERISK-29105:
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I am not sure what the other side refers to but do you suggest:

a.)
Inside chan_pjsip_indicate for condition==AST_CONTROL_RINGING as a result of another channel receiving the 180, we extend the check for pjmedia_sdp_neg_get_state() to cover this scenario.

b.) 
Inside chan_pjsip_incoming_response status.code==180 for the channel that received the SIP 180, we queue a AST_CONTROL_PROGRESS instead of the ringing?


And should we put this behind a configuration flag or do it unconditionally?

In case of a.) the neg_get_state could return   PJMEDIA_SDP_NEG_STATE_WAIT_NEGO and then pjsip will call sdp_neg_negotiate _before_ sending out the 183. Is this acceptable or should sdp_neg_negotiate be called somewhere inside Asterisk before queuing the AST_CONTROL_RINGING?

> 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.



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