[asterisk-bugs] [JIRA] (ASTERISK-29978) chan_sip: Asterisk do not use the first media format in reply with SDP

[Mark Petersen (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29978?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=258410#comment-258410 ] 

Mark Petersen edited comment on ASTERISK-29978 at 3/21/22 7:48 AM:
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SIP + RTP trace of a call with one way sound  on "209"
asterisk 18.10.1


was (Author: asterisk.org at zombie.dk):
SIP + RTP trace of a call with one way sound  on "209"

> chan_sip: Asterisk do not use the first media format in reply with SDP
> ----------------------------------------------------------------------
>
>                 Key: ASTERISK-29978
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29978
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Channels/chan_sip/CodecHandling
>    Affects Versions: 16.24.1, 18.10.1
>            Reporter: Mark Petersen
>         Attachments: Asterisk_debug.log, tcpdump-dev-asterisk.pcap
>
>
> when Asterisk receive 200 OK from Bob, it do not use the first codec in the 200 OK
> causing mitch match RTP, witch result in one way sound on most phones
> sip.conf
> disallow=all
> allow=g722,alaw,ulaw,gsm
> sip show peer hpbx 
>   Codecs       : (g722|alaw|ulaw|gsm)
> INVITE 
> m=audio 19534 RTP/AVP 8 9 0 3 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:3 GSM/8000
> a=rtpmap:101 telephone-event/8000
> 200 OK
> m=audio 51172 RTP/AVP 8 9 0 101
> a=rtpmap:8 PCMA/8000
> a=rtpmap:9 G722/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> asterisk send using rtpmap:9 G722/8000 
> but should be using rtpmap:8 PCMA/8000
> according ti RFC https://datatracker.ietf.org/doc/html/rfc3264#section-7



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