[asterisk-bugs] [JIRA] (ASTERISK-24122) res_pjsip mangles SAVPF invite if use_avpf=no

[James Van Vleet (JIRA)]

James Van Vleet created ASTERISK-24122:
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             Summary: res_pjsip mangles SAVPF invite if use_avpf=no
                 Key: ASTERISK-24122
                 URL: https://issues.asterisk.org/jira/browse/ASTERISK-24122
             Project: Asterisk
          Issue Type: Bug
      Security Level: None
          Components: Resources/res_pjsip
    Affects Versions: 12.4.0
            Reporter: James Van Vleet
            Severity: Minor


Originally testing 12.4.0-rc1.  Also tested with 12 head (r419442).  Both version of asterisk using pjproject commit 21b3576112c98556a9753e3048bac5ba5a84881a (newest at the time) from the asterisk fork on github.

Reference: http://svnview.digium.com/svn/asterisk?view=revision&revision=408502

It appears that use_avpf is broken assuming I understand the referenced commit above.

Testing using SIPML5 generated SDP:

v=0
o=- 5208809120103833000 2 IN IP4 127.0.0.1
s=Doubango Telecom - chrome
t=0 0
a=group:BUNDLE audio
a=msid-semantic: WMS 2VaiG8N52dPmzSpovitoTzRdJ5dfCosuriag
m=audio 37070 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 <SNIP>
a=rtcp:37070 IN IP4 <SNIP>
a=candidate:3424360271 1 udp 2122260223 <SNIP> 55001 typ host generation 0
a=candidate:3424360271 2 udp 2122260223 <SNIP> 55001 typ host generation 0
a=candidate:386226586 1 udp 2122194687 <SNIP> 37070 typ host generation 0
a=candidate:386226586 2 udp 2122194687 <SNIP> 37070 typ host generation 0
a=candidate:3683149321 1 udp 1685987071 <SNIP> 37070 typ srflx raddr 192.168.54.111 rport 37070 generation 0
a=candidate:3683149321 2 udp 1685987071 <SNIP> 37070 typ srflx raddr 192.168.54.111 rport 37070 generation 0
a=candidate:2191027135 1 tcp 1518280447 <SNIP> 0 typ host generation 0
a=candidate:2191027135 2 tcp 1518280447 <SNIP> 0 typ host generation 0
a=candidate:1501996394 1 tcp 1518214911 <SNIP> 0 typ host generation 0
a=candidate:1501996394 2 tcp 1518214911 <SNIP> 0 typ host generation 0
a=ice-ufrag:RjtUyTT8FJWTlMPf
a=ice-pwd:x9hiS9/Il/ADx+Zr0lTClxsG
a=ice-options:google-ice
a=fingerprint:sha-256 69:D0:0C:C3:A5:4F:C3:50:56:8F:49:4D:E8:DF:06:E7:08:70:2A:32:BC:F7:2D:00:CA:B3:EE:14:54:52:9E:9F
a=setup:actpass
a=mid:audio
a=extmap:1 urn:ietf:params:rtp-hdrext:ssrc-audio-level
a=extmap:3 http://www.webrtc.org/experiments/rtp-hdrext/abs-send-time
a=sendrecv
a=rtcp-mux
a=rtpmap:111 opus/48000/2
a=fmtp:111 minptime=10
a=rtpmap:103 ISAC/16000
a=rtpmap:104 ISAC/32000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:106 CN/32000
a=rtpmap:105 CN/16000
a=rtpmap:13 CN/8000
a=rtpmap:126 telephone-event/8000
a=maxptime:60
a=ssrc:4041464331 cname:T0+0+J3abDTx6xmg
a=ssrc:4041464331 msid:2VaiG8N52dPmzSpovitoTzRdJ5dfCosuriag babc4f16-bdfe-4081-bb35-e73eba8302ca
a=ssrc:4041464331 mslabel:2VaiG8N52dPmzSpovitoTzRdJ5dfCosuriag
a=ssrc:4041464331 label:babc4f16-bdfe-4081-bb35-e73eba8302ca

use_avpf=yes for the endpoint in question produces a good SDP response with negotiated ice information, media and the like.  All is well.  However with use_avpf=no the following odd thing is sent:

Content-Type: application/sdp
Content-Length:   173

v=0
o=- 4008842760 4 IN IP4 <SNIP>
s=Asterisk
c=IN IP4 <SNIP>
t=0 0
m=audio 0 UDP/TLS/RTP/SAVPF 111 103 104 0 8 106 105 13 126
c=IN IP4 <SNIP>

<next log line here>

For some reason all ICE and media information is missing from the response SDP.  Notice the Content-Length shows the abbreviated SDP is at least intentional at that level.

This does not happen with all call types - calls using blink to the same endpoint (no encryption if I recall) work fine however the the sipML5 phone is constantly broken.

I don't think it matters but I should note that we are not using the asterisk web-socket interfaces at the moment - that is terminating on a different gateway and the sdp forwarded to asterisk.

There is nothing different (interesting) in the debug log between use_avpf yes and no however I can attach if necessary.

[endpoint]
type=endpoint
context=default                 ; Default context for incoming calls
allow=ulaw
direct_media=no
rtp_symmetric=yes
media_address=54.187.64.133
use_avpf=yes ;# asterisk will accept avfp if this is false.  this *requires* avpf
ice_support=yes
media_encryption=dtls
dtls_cert_file=/etc/asterisk/certs/fakedomain.com.crt     
dtls_private_key=/etc/asterisk/certs/fakedomain.com.key  

(with blink I also comment out media_encryption)





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