[asterisk-bugs] [JIRA] (ASTERISK-29109) Asterisk 18 does not progress calls due to codec negotiation after upgrading from Asterisk 16
Ross Beer (JIRA)
noreply at issues.asterisk.org
Tue Oct 6 08:18:36 CDT 2020
[ https://issues.asterisk.org/jira/browse/ASTERISK-29109?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]
Ross Beer updated ASTERISK-29109:
---------------------------------
Description:
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:
{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:
{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}
was:
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}
> 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
> Affects Versions: 18.0.0
> Environment: CentOS 7
> Reporter: Ross Beer
> 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:
> {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:
> {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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