Elastix Tandberg Video support H.264

Discussion in 'General' started by gogasca, Nov 20, 2010.

  1. gogasca

    Joined:
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    I have 2 Tandberg E20s that video works between each other but...
    quality is negotiated to lower one (QCIF).
    I see that Elastix removes fmtp from SDP messages

    768 x 448@30fps (w448p)
    576 x 448@30fps (448p)
    512 x 288@30fps (w288p)
    352 x 288@30fps (CIF)
    176 x 144@30fps (QCIF)

    I see that when we get the INVITE with SDP Elastix removes fmtp values when forwarding INVITE to other peer.

    INVITE sip:103@110.10.0.200 SIP/2.0
    Via: SIP/2.0/UDP 110.10.0.144:5060;branch=z9hG4bK90b3d0e51cfdbe6e4602a611dcfd5918.1;rport
    Call-ID: 201764a6216109dd@110.10.0.144
    CSeq: 101 INVITE
    Contact: <sip:104@110.10.0.144:5060>
    From: <sip:104@110.10.0.200>;tag=5420a486a80dbc77
    To: <sip:103@110.10.0.200>
    Max-Forwards: 70
    Route: <sip:110.10.0.200;lr>
    Allow: INVITE,ACK,CANCEL,BYE,UPDATE,INFO,OPTIONS,REFER,NOTIFY
    User-Agent: TANDBERG/257 (TE2.2.1.224666)
    Authorization: Digest nonce="2b87ef6a", realm="asterisk", username="104", uri="sip:110.10.0.200", response="f2c6901e89bf6825116c03e4822c7009", algorithm=MD5
    Supported: replaces,100rel,timer,gruu,path,outbound
    Session-Expires: 500
    Content-Type: application/sdp
    Content-Length: 1225

    v=0
    o=tandberg 4 1 IN IP4 110.10.0.144
    s=-
    c=IN IP4 110.10.0.144
    b=CT:768
    t=0 0
    m=audio 2334 RTP/AVP 100 101 102 9 18 11 8 0 103
    c=IN IP4 110.10.0.144
    b=TIAS:64000
    a=rtpmap:100 MP4A-LATM/90000
    a=fmtp:100 profile-level-id=24;object=23;bitrate=64000
    a=rtpmap:101 G7221/16000
    a=fmtp:101 bitrate=32000
    a=rtpmap:102 G7221/16000
    a=fmtp:102 bitrate=24000
    a=rtpmap:9 G722/8000
    a=rtpmap:18 G729/8000
    a=fmtp:18 annexb=yes
    a=rtpmap:11 L16/16000
    a=rtpmap:8 PCMA/8000
    a=rtpmap:0 PCMU/8000
    a=rtpmap:103 telephone-event/8000
    a=fmtp:103 0-15
    a=sendrecv
    m=video 2336 RTP/AVP 97 98 99 34 31
    b=TIAS:768000
    a=rtpmap:97 H264/90000
    a=fmtp:97 profile-level-id=42800d;max-mbps=40500;max-fs=1344;max-smbps=40500
    a=rtpmap:98 H264/90000
    a=fmtp:98 profile-level-id=42800d;max-mbps=40500;max-fs=1344;max-smbps=40500;packetization-mode=1
    a=rtpmap:99 H263-1998/90000
    a=fmtp:99 custom=1024,768,4;custom=1024,576,4;custom=800,600,4;cif4=2;custom=720,480,2;custom=640,480,2;custom=512,288,1;cif=1;custom=352,240,1;qcif=1;maxbr=7680
    a=rtpmap:34 H263/90000
    a=fmtp:34 cif=1;qcif=1;maxbr=7680
    a=rtpmap:31 H261/90000
    a=fmtp:31 cif=1;qcif=1;maxbr=7680
    a=rtcp-fb:* nack pli
    a=sendrecv
    a=content:main
    a=label:11
    a=answer:full


    Found RTP video format 97
    Found RTP video format 98
    Found RTP video format 99
    Found RTP video format 34
    Found RTP video format 31
    Found video description format H264 for ID 97
    Found video description format H264 for ID 98
    Found video description format H263-1998 for ID 99
    Found video description format H263 for ID 34
    Found video description format H261 for ID 31

    ===

    Audio is at 110.10.0.200 port 17922
    Video is at 110.10.0.200 port 14696
    Adding codec 0x4 (ulaw) to SDP
    Adding codec 0x1000 (g722) to SDP
    Adding video codec 0x200000 (h264) to SDP
    Adding non-codec 0x1 (telephone-event) to SDP


    Reliably Transmitting (NAT) to 110.10.0.136:5060:
    INVITE sip:103@110.10.0.136:5060 SIP/2.0
    Via: SIP/2.0/UDP 110.10.0.200:5060;branch=z9hG4bK3f194b88;rport
    Max-Forwards: 70
    From: "Mr T" <sip:104@110.10.0.200>;tag=as21c38e6e
    To: <sip:103@110.10.0.136:5060>
    Contact: <sip:104@110.10.0.200>
    Call-ID: 456556eb59a5c74b57ccb1461e79ead1@110.10.0.200
    CSeq: 102 INVITE
    User-Agent: Asterisk PBX 1.6.2.10
    Date: Sat, 20 Nov 2010 06:15:57 GMT
    Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
    Supported: replaces, timer
    Content-Type: application/sdp
    Content-Length: 330

    v=0
    o=root 29229991 29229991 IN IP4 110.10.0.200
    s=Asterisk PBX 1.6.2.10
    c=IN IP4 110.10.0.200
    b=CT:1080
    t=0 0
    m=audio 17922 RTP/AVP 0 9 103
    a=rtpmap:0 PCMU/8000
    a=rtpmap:9 G722/8000
    a=rtpmap:103 telephone-event/8000
    a=fmtp:103 0-16
    a=ptime:20
    a=sendrecv
    m=video 14696 RTP/AVP 97
    a=rtpmap:97 H264/90000
    a=sendrecv


    Any idea?
    How can modify chan_sip.c ?

    http://lists.digium.com/pipermail/aster ... 02727.html

    Thanks
     

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