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

[Andrew Nagy (JIRA)]

Andrew Nagy created ASTERISK-24835:
--------------------------------------

             Summary: 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


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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