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

[Mark Petersen (JIRA)]

Mark Petersen created ASTERISK-29978:
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             Summary: 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: 18.10.1, 16.24.1
            Reporter: Mark Petersen
         Attachments: 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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