FreePBX removes TIAS

Hi community,

we have setup two trunks: TRUNK-IN (chan_sip) & TRUNK-OUT (chan_pjsip) on FreePBX.

Our PJSUA client | Alice (172.21.X.X) makes a video call towards Bob (10.40.Z.Z) over FreePBX (172.21.Y.Y).
→ Call path: Alice → FreePBX TRUNK-IN → FreePBX TRUNK-OUT → Bob.

The SDP offer (INVITE) from PJSUA client:

    Request-Line: INVITE sip:[email protected] SIP/2.0
    Message Header
    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 3859967387 3859967387 IN IP4 172.21.X.X
            Session Name (s): pjmedia
            Bandwidth Information (b): AS:6417
            Time Description, active time (t): 0 0
            Session Attribute (a): X-nat:0
            Media Description, name and address (m): audio 19900 RTP/AVP 8 0 9 96 120 121
            Connection Information (c): IN IP4 172.21.X.X
            **Bandwidth Information (b): TIAS:96000**
            Media Attribute (a): rtcp:19901 IN IP4 172.21.X.X
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:8 PCMA/8000
            Media Attribute (a): rtpmap:0 PCMU/8000
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): rtpmap:96 opus/48000/2
            Media Attribute (a): fmtp:96 useinbandfec=1
            Media Attribute (a): rtpmap:120 telephone-event/8000
            Media Attribute (a): fmtp:120 0-16
            Media Attribute (a): rtpmap:121 telephone-event/48000
            Media Attribute (a): fmtp:121 0-16
            Media Attribute (a): ssrc:1336894421 cname:352078195d8c8c18
            Media Description, name and address (m): video 19902 RTP/AVP 97 102
            Connection Information (c): IN IP4 172.21.X.X
            **Bandwidth Information (b): TIAS:6000000**
            Media Attribute (a): rtcp:19903 IN IP4 172.21.X.X
            Media Attribute (a): sendrecv
            Media Attribute (a): rtpmap:97 H264/90000
            Media Attribute (a): fmtp:97 profile-level-id=42801F;packetization-mode=1;max-br=5000;max-fs=8100;max-mbps=490000;max-fps=3000
            Media Attribute (a): rtpmap:102 VP8/90000
            Media Attribute (a): fmtp:102 max-fr=30; max-fs=580
            Media Attribute (a): ssrc:1866030669 cname:352078195d8c8c18
            Media Attribute (a): rtcp-fb:* nack pli
            [Generated Call-ID: d808d070-2459-4993-8fe9-2dd2497a69e5]

The SDP offer (INVITE) from FreePBX to Bob:

    Message Body
        Session Description Protocol
            Session Description Protocol Version (v): 0
            Owner/Creator, Session Id (o): - 1811435004 1811435004 IN IP4 172.21.Y.Y
            Session Name (s): Asterisk
            Connection Information (c): IN IP4 172.21.Y.Y
            Time Description, active time (t): 0 0
            Media Description, name and address (m): audio 14986 RTP/AVP 107 9 0 8 101
            Media Attribute (a): rtpmap:107 opus/48000/2
            Media Attribute (a): fmtp:107 useinbandfec=1
            Media Attribute (a): rtpmap:9 G722/8000
            Media Attribute (a): rtpmap:0 PCMU/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:20
            Media Attribute (a): sendrecv
            Media Description, name and address (m): video 13178 RTP/AVP 99
            Media Attribute (a): rtpmap:99 H264/90000
            Media Attribute (a): fmtp:99 max-mbps=490000;max-fs=8100;max-br=5000;max-fps=3000;packetization-mode=1;profile-level-id=42801F
            Media Attribute (a): sendrecv
            [Generated Call-ID: 82e9833f-d2c3-4a46-a0d3-1b23b8e84506]

FreePBX removes the “Bandwidth Information”, TIAS, attributes from the SDP offer.

Now that this information is not relayed to Bob, it sets the rate of video that it sends to Alice to a very low level (resolution).

Any idea about why FreePBX removes the bandwidth information from SDP and how to put it back?

Maximum bitrate for video calls is set to 6000 kbps on FreePBX (advanced SIP settings).

FreePBX version: FPBX-15.0.17.67(16.17.0)

Thanks in advance.

It’s not a SIP proxy so information isn’t just proxied through such as TIAS. Neither chan_sip or chan_pjsip know about or place such an attribute in their SDP, doing so would require modifying the code.

Any plans to modify the code anytime soon?

Thanks.

I know of noone working on such a thing.

Doing so in a general way would likely be a major piece of work.

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