Samuel,

Thank you for the patch. We will test and commit it.

On 6/17/2010 8:18 AM, samuel.engelma...@thalesgroup.com wrote:
> Patch to be applied on rtpproxy-1.2.1 notls
> *********************************
> diff -urp ../rtpproxy-1.2.1/rtp_resizer.c ./rtp_resizer.c
> --- ../rtpproxy-1.2.1/rtp_resizer.c 2009-01-12 12:36:40.000000000 +0100
> +++ ./rtp_resizer.c 2010-06-17 14:20:30.000000000 +0200
> @@ -245,7 +245,7 @@ rtp_resizer_get(struct rtp_resizer *this
>         ret = rtp_packet_alloc();
>         if (ret == NULL)
>      break;
> -      memcpy(ret, p, offsetof(struct rtp_packet, data.buf));
> +      memcpy(ret, p, offsetof(struct rtp_packet, data.buf) +
> sizeof(rtp_hdr_t));
>         move_chunk(ret, p, &chunk);
>         ++split;
>     }
> *******************************
> Fixes the bug described above:
> /Hi all./
> //
> /I am having trouble with repacking. /
> //
> /Occurs when :/
> /- target packet size is smaller than received size, e.g : received
> packets are 120ms, repack size is 30ms/
> /AND/
> /- " received_size / target_size " is a round integer: 120/30 = 4, or
> 100/20=5 , etc.../
> //
> /I have no problem with ratio like 100/60 or 100/40, that is to
> say, anytime their is a residual chunck, it works fine./
> //
> /Symptoms are:/
> /- all the repacked packets are incorrect except the last one, e.g when
> 120ms->30ms : repacking generates three incorrect 30ms packet followed
> by the last 30ms - correct./
> /- incorrect packets have a correct IP/UDP part but RTP header and RTP
> payload is all zero or nothing at all/
> //
> /Is there any workaround for this problem? /
> /Or am I doing something wrong ?/
> //
> /rtpproxy is running in bridge mode, driven by nathelper / openSips,
> using force_trp_proxy( ) with the "zNN" option /
> //
> /Occurs with 723.1 and also 729 codecs./
> **********
> Reason was "move_chunck()"  did not memcpy the rtp_hdr_t from src chunck
> to newly allocated rtp packet
> Following repacks now work well:
>   - G729
>      100ms into 20ms
>      200ms into 10ms
> -G723.1
>      120ms into 60ms
>      180ms into 30ms
> Still to be tested on extra codec (711, etc).
> Please let me know if you have further results.

Regards,
-- 
Maksym Sobolyev
Sippy Software, Inc.
Internet Telephony (VoIP) Experts
T/F: +1-646-651-1110
Web: http://www.sippysoft.com
MSN: sa...@sippysoft.com
Skype: SippySoft

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