[asterisk-bugs] [JIRA] (ASTERISK-29109) res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16

[Friendly Automation (JIRA)]

    [ https://issues.asterisk.org/jira/browse/ASTERISK-29109?page=com.atlassian.jira.plugin.system.issuetabpanels:comment-tabpanel&focusedCommentId=252411#comment-252411 ] 

Friendly Automation commented on ASTERISK-29109:
------------------------------------------------

Change 15040 merged by Joshua Colp:
res_pjsip: Adjust outgoing offer call pref.

[https://gerrit.asterisk.org/c/asterisk/+/15040|https://gerrit.asterisk.org/c/asterisk/+/15040]

> res_pjsip_session: Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16
> ----------------------------------------------------------------------------------------------------------------
>
>                 Key: ASTERISK-29109
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-29109
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip_session
>    Affects Versions: 18.0.0
>         Environment: CentOS 7
>            Reporter: Ross Beer
>            Assignee: Joshua C. Colp
>            Severity: Minor
>
> If you have a phone using an endpoint only allowing 'g722' and attempt a call to a second endpoint (trunk) that has 'alaw, g722' but the far end answering the call only supports 'alaw' the call ends with '488 Unsupported Media Type'.
> This is due to the outgoing SIP packet only contains one codec, 'g722', which is the common codec between the phones endpoint configuration and the trunk endpoint configuration.
> {noformat}
> [phone]
> disallow=all
> allow=g722
> [trunk]
> disallow=all
> allow=alaw,g722
> {noformat}
> Outgoing SIP packet Asterisk 18:
> {noformat}
> Session Initiation Protocol (INVITE)
>     Request-Line: INVITE sip:+<TEL>@<IP>:5060 SIP/2.0
>     Message Header
>     Message Body
>         Session Description Protocol
>             Session Description Protocol Version (v): 0
>             Owner/Creator, Session Id (o): - 1534344650 1534344650 IN IP4 <IP>
>             Session Name (s): Asterisk
>             Connection Information (c): IN IP4 <IP>
>             Time Description, active time (t): 0 0
>             Media Description, name and address (m): audio 22170 RTP/AVP 9 101
>             Media Attribute (a): rtpmap:9 G722/8000
>             Media Attribute (a): rtpmap:101 telephone-event/8000
>             Media Attribute (a): fmtp:101 0-16
>             Media Attribute (a): ptime:20
>             Media Attribute (a): maxptime:150
>             Media Attribute (a): sendrecv
>             [Generated Call-ID: 9e176b7f-cb95-4c64-9ce2-59f41c94997d]
> {noformat}
> When using the same configuration via Asterisk 16, asterisk passes both codecs on the trunk call.
> Outgoing SIP packet Asterisk 16:
> {noformat}
> Session Initiation Protocol (INVITE)
>     Request-Line: INVITE sip:+<TEL>@<IP>:5060 SIP/2.0
>     Message Header
>     Message Body
>         Session Description Protocol
>             Session Description Protocol Version (v): 0
>             Owner/Creator, Session Id (o): - 1351849362 1351849362 IN IP4 <IP>
>             Session Name (s): Asterisk
>             Connection Information (c): IN IP4 <IP>
>             Time Description, active time (t): 0 0
>             Media Description, name and address (m): audio 19714 RTP/AVP 9 8 101
>             Media Attribute (a): rtpmap:9 G722/8000
>             Media Attribute (a): rtpmap:8 PCMA/8000
>             Media Attribute (a): rtpmap:101 telephone-event/8000
>             Media Attribute (a): fmtp:101 0-16
>             Media Attribute (a): ptime:20
>             Media Attribute (a): maxptime:150
>             Media Attribute (a): sendrecv
>             [Generated Call-ID: b8e5a1b8-f507-48e6-9bc4-67ea255347f3]
> {noformat}



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