[asterisk-bugs] [JIRA] (ASTERISK-27702) allow video/h264 breaks symmetric rtp

[Jørgen H (JIRA)]

     [ https://issues.asterisk.org/jira/browse/ASTERISK-27702?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel ]

Jørgen H updated ASTERISK-27702:
--------------------------------

    Description: 
Adding codec h264 in allow-list result in that rtp_symmetric gets ignored and asterisk sends rtp audio to the ip-address in the sdp even when h264 is not in use.

Config:
rewrite_contact = yes
rtp_symmetric = yes
force_rport = yes
direct_media = no
allow = !all,opus,g722,alaw,ulaw,h264

SDP from answering telephone when audio works without h264 in allow list:

v=0
o=MxSIP 0 1 IN IP4 10.24.5.115
s=SIP Call
c=IN IP4 10.24.5.115
t=0 0
m=audio 3000 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

SDP when h264 is enabled and audio does not work (audio is sent to 10.24.5.115):

v=0
o=MxSIP 0 1 IN IP4 10.24.5.115
s=SIP Call
c=IN IP4 10.24.5.115
t=0 0
m=audio 3000 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99



  was:
Adding codec h264 in allow-list result in that rtp_symmetric gets ignored and asterisk sends rtp auto to the ip-address in the sdp even when h264 is not in use.

Config:
rewrite_contact = yes
rtp_symmetric = yes
force_rport = yes
direct_media = no
allow = !all,opus,g722,alaw,ulaw,h264

SDP from answering telephone when audio works without h264 in allow list:

v=0
o=MxSIP 0 1 IN IP4 10.24.5.115
s=SIP Call
c=IN IP4 10.24.5.115
t=0 0
m=audio 3000 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv

SDP when h264 is enabled and audio does not work (audio is sent to 10.24.5.115):

v=0
o=MxSIP 0 1 IN IP4 10.24.5.115
s=SIP Call
c=IN IP4 10.24.5.115
t=0 0
m=audio 3000 RTP/AVP 9 8 0 101
a=rtpmap:9 G722/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=silenceSupp:off - - - -
a=fmtp:101 0-15
a=ptime:20
a=sendrecv
m=video 0 RTP/AVP 99




> allow video/h264 breaks symmetric rtp
> -------------------------------------
>
>                 Key: ASTERISK-27702
>                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-27702
>             Project: Asterisk
>          Issue Type: Bug
>      Security Level: None
>          Components: Resources/res_pjsip
>    Affects Versions: 14.5.0, 15.2.2
>         Environment: linux x64
> Verified on 
> asterisk 15.2.2 with pjsip 2.7.2
> asterisk 14.5 and pjsip 2.6
>            Reporter: Jørgen H
>              Labels: pjsip
>
> Adding codec h264 in allow-list result in that rtp_symmetric gets ignored and asterisk sends rtp audio to the ip-address in the sdp even when h264 is not in use.
> Config:
> rewrite_contact = yes
> rtp_symmetric = yes
> force_rport = yes
> direct_media = no
> allow = !all,opus,g722,alaw,ulaw,h264
> SDP from answering telephone when audio works without h264 in allow list:
> v=0
> o=MxSIP 0 1 IN IP4 10.24.5.115
> s=SIP Call
> c=IN IP4 10.24.5.115
> t=0 0
> m=audio 3000 RTP/AVP 9 8 0 101
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=silenceSupp:off - - - -
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> SDP when h264 is enabled and audio does not work (audio is sent to 10.24.5.115):
> v=0
> o=MxSIP 0 1 IN IP4 10.24.5.115
> s=SIP Call
> c=IN IP4 10.24.5.115
> t=0 0
> m=audio 3000 RTP/AVP 9 8 0 101
> a=rtpmap:9 G722/8000
> a=rtpmap:8 PCMA/8000
> a=rtpmap:0 PCMU/8000
> a=rtpmap:101 telephone-event/8000
> a=silenceSupp:off - - - -
> a=fmtp:101 0-15
> a=ptime:20
> a=sendrecv
> m=video 0 RTP/AVP 99



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