[asterisk-bugs] [JIRA] (ASTERISK-24835) Early Media Not working with Chan SIP and Asterisk 13

[Andrew Nagy (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-24835?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Andrew Nagy updated ASTERISK-24835:
-----------------------------------

    Attachment: dialplan.txt

> Early Media Not working with Chan SIP and Asterisk 13
> -----------------------------------------------------
>
>                 Key: ASTERISK-24835
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24835
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>    Affects Versions: 13.1.1
>            Reporter: Andrew Nagy
>            Severity: Critical
>         Attachments: chanpjsip.txt, chansip.txt, dialplan.txt
>
>
> Early media is not working in Asterisk 13 when using Chan_SIP. It works fine when using PJSIP. The reasoning appears to be that Chan_SIP is sending 180 ringing when it's told to "Progress" while PJSIP is sending 183 when it's told to progress.
> I have attached a log of a chan_sip dial and a pjsip dial. You will see the differences there in. Specifically:
> CHAN_SIP:
> {code}
>     -- Executing [999 at from-internal:3] Progress("SIP/1002-00000085", "") in new stack
> <--- Transmitting (NAT) to 66.185.28.100:5064 --->
> SIP/2.0 180 Ringing
> Via: SIP/2.0/UDP 10.10.10.12:5064;branch=z9hG4bK2113418340;received=66.185.28.100;rport=5064
> From: <sip:1002 at freepbxdev1.schmoozecom.net:5061>;tag=706301242
> To: <sip:999 at freepbxdev1.schmoozecom.net:5061>;tag=as3db28b80
> Call-ID: 2099708635-5064-5 at BA.BA.BA.BC
> CSeq: 41 INVITE
> Server: FPBX-13.0.1alpha1(13.1.1)
> Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE
> Supported: replaces, timer
> Session-Expires: 1800;refresher=uas
> Contact: <sip:999 at 199.102.239.103:5061>
> Content-Length: 0
> <------------>
> {code}
> CHAN_PJSIP:
> {code}
>     -- Executing [999 at from-internal:3] Progress("PJSIP/1001-000000aa", "") in new stack
> <--- Transmitting SIP response (827 bytes) to UDP:66.185.28.100:5060 --->
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 10.10.10.12:5060;rport=5060;received=66.185.28.100;branch=z9hG4bK503980922
> Call-ID: 70708105-5060-5 at BA.BA.BA.BC
> From: "Andrew Nagy" <sip:1001 at freepbxdev1.schmoozecom.net>;tag=1590118380
> To: <sip:999 at freepbxdev1.schmoozecom.net>;tag=AvwkMijOGvO1QXW1qDcNXl8xlBqUW.Ux
> CSeq: 41 INVITE
> Server: FPBX-13.0.1alpha1(13.1.1)
> Contact: <sip:199.102.239.103:5065>
> Allow: OPTIONS, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, MESSAGE, REGISTER, REFER
> Content-Type: application/sdp
> Content-Length:   255
> v=0
> o=- 8004 8002 IN IP4 199.102.239.103
> s=Asterisk
> c=IN IP4 199.102.239.103
> t=0 0
> m=audio 14388 RTP/AVP 0 8 101
> a=rtpmap:0 PCMU/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:101 telephone-event/8000
> a=fmtp:101 0-16
> a=ptime:20
> a=maxptime:150
> a=sendrecv
> {code}



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