[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:all-tabpanel ]

Mark Petersen updated ASTERISK-29978:
-------------------------------------

    Description: 
Alice call Bob
Allice support alaw,ulaw Bob support g722,alaw,ulaw
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

  was:
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


> 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
>
>
> Alice call Bob
> Allice support alaw,ulaw Bob support g722,alaw,ulaw
> 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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