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

[Sven Andersen (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29105?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=255611#comment-255611 ] 

Sven Andersen commented on ASTERISK-29105:
------------------------------------------

This change seems to be problematic in some situations. I digged a "no ringback-tone"-issue down to this change, reverting fixed it. 

Calling T-Mobile germany from o2 germany _sometimes_ (~1 out of 5 times) responds with 180/SDP but actually does not send the expected/announced audio-stream. So there is no indication that the remote party is ringing and suddenly the call is established (If the caller waits long enough and does not hang up due to missing ringback). 

Could this be made configurable? Otherwise I would need to patch every new release for myself.

Should I try and submit an according patch (though I am not an asterisk/SIP developer or c-expert so don't expect it to be perfect...)?

> 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
>      Target Release: 16.17.0, 18.3.0
>
>         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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