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

Thu Feb 26 20:29:34 CST 2015

    [ https://issues.asterisk.org/jira/browse/ASTERISK-24835?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=225187#comment-225187 ] 

Andrew Nagy commented on ASTERISK-24835:
----------------------------------------

And then I found the actual Review Board request that started all of this (https://reviewboard.asterisk.org/r/3700/diff/).

So my question here now is, why with "defaults" does chan_sip send a 180 but pjsip sends a 183. I think they should either both send a 180 or a 183 when everything is defaulted to it's minimalistic state because right now you have different working characteristics between the two drivers. (one needs an option set to use early media in the dialplan, the other does not)

> 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
>          Components: Channels/chan_sip/General
>    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 (13):
> {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 (13):
> {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}
> CHAN_SIP(11.16.0)
> {code}
>     -- Executing [999 at from-internal:3] Progress("SIP/1002-00000001", "") in new stack
> Audio is at 12696
> Adding codec 100003 (ulaw) to SDP
> Adding codec 100004 (alaw) to SDP
> Adding non-codec 0x1 (telephone-event) to SDP
> <--- Transmitting (NAT) to 66.185.28.100:5064 --->
> SIP/2.0 183 Session Progress
> Via: SIP/2.0/UDP 10.10.10.12:5064;branch=z9hG4bK1753731864;received=66.185.28.100;rport=5064
> From: <sip:1002 at freepbxdev1.schmoozecom.net:5061>;tag=9700035
> To: <sip:999 at freepbxdev1.schmoozecom.net:5061>;tag=as5bd4933a
> Call-ID: 1953294974-5064-7 at BA.BA.BA.BC
> CSeq: 61 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-Type: application/sdp
> Require: timer
> Content-Length: 264
> v=0
> o=root 424885368 424885368 IN IP4 199.102.239.103
> s=Asterisk PBX 11.16.0
> c=IN IP4 199.102.239.103
> t=0 0
> m=audio 12696 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=sendrecv
> <------------>
> {code}
> The above logs were generated from the dialplan below
> {code}
> [bad-number]
> include => bad-number-custom
> exten => _X.,1,ResetCDR()
> exten => _X.,n,NoCDR()
> exten => _X.,n,Progress
> exten => _X.,n,Wait(1)
> exten => _X.,n,Progress
> exten => _X.,n,Playback(silence/1&cannot-complete-as-dialed&check-number-dial-again,noanswer)
> exten => _X.,n,Wait(1)
> exten => _X.,n,Congestion(20)
> exten => _X.,n,Hangup
> {code}
> I have, however, attached a simplified dialplan for testing in the attachments.

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