[asterisk-bugs] [JIRA] (ASTERISK-20976) Asterisk 11.2.1 RTP packetization setting ignored in RTP output

[i2045 (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-20976?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

i2045 updated ASTERISK-20976:
-----------------------------

    Attachment: Asterisk_11_2_1_RTP_packetization.pcap
    
> Asterisk 11.2.1 RTP packetization setting ignored in RTP output
> ---------------------------------------------------------------
>
>                 Key: ASTERISK-20976
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-20976
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: . I did not set the category correctly.
>    Affects Versions: 11.2.1
>         Environment: Ubuntu 12.04 LTS server
>            Reporter: i2045
>         Attachments: Asterisk_11_2_1_RTP_packetization.pcap
>
>
> The setup:
> 192.168.178.250 Endpoint: Zoiper softphone (initiates the call)
> 192.168.178.31 Endpoint: Asterisk (answerers the call)
> 192.168.178.30 Asterisk 11.2.1 (between Zoiper and other Asterisk) + make samples + these changes:
> sip.conf
> 114a115
> > canreinvite=no
> 309,310c310,311
> < ;disallow=all                   ; First disallow all codecs
> < ;allow=ulaw                     ; Allow codecs in order of preference
> ---
> > disallow=all                   ; First disallow all codecs
> > allow=ulaw:30                     ; Allow codecs in order of preference
> extensions.conf
> 656a657
> > exten => 666,1,dial(SIP/${EXTEN}@192.168.178.31)
> Zoiper calls .30 (Asterisk 11.2.1 box in the middle).
> .30 dials .31.
> .31 answerers the call and plays the demo-congrats.
> Now Asterisk 11.2.1 should send RTP packets each 30ms to Zoiper.
> This is also announced in the signalling:
> U 2013/01/23 15:50:08.419046 192.168.178.30:5060 -> 192.168.178.250:5060
> SIP/2.0 200 OK.
> ...
> v=0.
> o=root 983949886 983949886 IN IP4 192.168.178.30.
> s=Asterisk PBX 11.2.1.
> c=IN IP4 192.168.178.30.
> t=0 0.
> m=audio 13074 RTP/AVP 0 101.
> a=rtpmap:0 PCMU/8000.
> a=rtpmap:101 telephone-event/8000.
> a=fmtp:101 0-16.
> a=silenceSupp:off - - - -.
> a=ptime:30.
> a=sendrecv.
> But the RTP packets are send each 20ms not 30ms.
> I have a packet capture attached.
> Is the RTP packetization setting supported in Asterisk 11?

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