Payload size change

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Hi Lucas,

I don't think that ptime attribute is to be negotiated, please see:

RFC 3264 Section 5.1:
   If the ptime attribute is present for a stream, it indicates the
   desired packetization interval that the offerer would like to
   receive.

RFC 3264 Section 6.1:
   The answerer MAY include a non-zero ptime attribute for any media
   stream; this indicates the packetization interval that the answerer
   would like to receive.  There is no requirement that the
   packetization interval be the same in each direction for a particular
   stream.


BR,
nanang


On Fri, Oct 2, 2009 at 6:38 PM, Lucas Tehbing <lucas_mvoip at hotmail.com> wrote:
> I added following code to set ptime in SDP
>
> pjmedia_sdp_attr
>
> *a;
>
> pj_str_t value;
>
> pjsua_var.p
>
> value = pj_str(
>
> "40");
>
> a = pjmedia_sdp_attr_create(pool,
>
> "ptime", &value); pjmedia_sdp_attr_add(&sdp->attr_count, sdp->attr, a);
>
> This added ptime as 40 in INVITE SDP but in Session Progress and 200 OK i
> receive ptime as 30. Does that mean ptime is being negotiated?
>
> Benny can you comment on this?
>
> Cheers ,
>
> Lucas
>
>
>
> ________________________________
> Date: Fri, 2 Oct 2009 13:21:32 +0200
> From: tal.fromm@xxxxxxxxx
> To: pjsip at lists.pjsip.org
> Subject: Re: Payload size change
>
> I had the same problem with other codec speex/8000.
> The server that connects the call will send packet size as it's default
> value, but if both client side are set to the requested payload size, and
> the server only passes the packet between them you should be fine.
> In other post I saw that for now it was selected not to send the payload due
> to different codec support. I'm still learning how to add attribute to the
> message, and then I will be able to send the ptime setting. Of course the
> other side must support SDP message and work with it.
> If you know how to do that you can try it, and please share the info with
> me.
> Good luck,
> Tal
> On Fri, Oct 2, 2009 at 7:44 AM, Lucas Tehbing <lucas_mvoip at hotmail.com>
> wrote:
>
> Sorry for getting back late.
> I have verified this. I configured codec ptime as 40 and compared packet
> sizes (g729). Complete outgoing packet is 52 (12+40) which is correct. But
> incoming packet is still 32 Bytes (12+20) this indicates that end callee
> does not know it has to send 40 Byre RTP payload.
>
> In this case there was no Ptime in SDP. I had seen Ptime in dump before but
> now i do not see it.
>
> Another doubt arises (i checked old posts on this matter also). Codec ptime
> is a codec property i think but in pjsua its public.
>
> Is there any way to make sure end callee can be conveyed that it will use
> different ptime?
>
> How can we change it at codec level direcly?
>
> Cheers Lucas
>
>
>
>
> ________________________________
> Date: Mon, 28 Sep 2009 21:42:54 +0200
> From: tal.fromm@xxxxxxxxx
> To: pjsip at lists.pjsip.org
> Subject: Re: Payload size change
>
> G729 is compressing 10msec voice to 80bits of data.
> I don't use G729, but if you found a way to set the SDP "ptime:20" value to
> be "ptime:40" and you set your configuration ptime=40 than it should work.
> I tried it with speex, setting ptime=100 (5 payload in one RTP), but I
> didn't see SDP with value "ptime:100".
> The value you set to ptime should be multiply of the basic value the coder
> support.
> In G729 it should be x10ms, but it could be that the codec implementation,
> in this case rap the data as?20ms.
> Hope it helped you, and if you can direct me to the SDP setting it would be
> great.
>
> On Mon, Sep 28, 2009 at 2:38 PM, Lucas Tehbing <lucas_mvoip at hotmail.com>
> wrote:
>
> There is one more point. I was reading APS documentation, they sayg729 data
> is sampled?for 10 ms. If we increase RTP payload does that mean g729 output
> should also be 20 ms ? or RTP payload can be increased with existing
> settings only.
>
> cheer,
>
> Lucas
>
> ________________________________
> From: lucas_mvoip@xxxxxxxxxxx
> To: pjsip at lists.pjsip.org
> Date: Sun, 27 Sep 2009 09:15:32 +0100
> Subject: Re: Payload size change
>
> Yes that is what i want to try. Two?payloads in one RTP. Theoritically we
> can two three 4 any amount of payload in one RTP packet right?
> In the Invite i see one SDP parameter ptime:20.?Is this the way it can tell
> other endpoint about payload?or it sends. or it is done using frame header
> of g729?
>
>
> ________________________________
> Date: Sun, 27 Sep 2009 07:40:00 +0200
> From: tal.fromm@xxxxxxxxx
> To: pjsip at lists.pjsip.org
> Subject: Re: Payload size change
>
> If I understand your question you want to get 2 payload packets sent in one
> RTP.
> You can do that with setting ptime=40.
> The problem is that as far as I saw pjsip doesn't send this info to the
> other side, and it the other side is not set to the same packet size you
> won't hear good.
>
> On Sat, Sep 26, 2009 at 5:52 PM, Manoj Joshi <manoj at ascenttelecom.com>
> wrote:
>
> Any comment?
>
> -----Original Message-----
> From: pjsip-bounces@xxxxxxxxxxxxxxx [mailto:pjsip-bounces at lists.pjsip.org]On
> Behalf Of Lucas Tehbing
> Sent: Saturday, September 26, 2009 3:21 AM
> To: pjsip at lists.pjsip.org
> Subject: Re: Payload size change
>
> Hello,
>
> I am testing pjsip symbian APS-Direct. I want to change RTP Payload size
> when using Passthrough g729 from 20 to 40. Where would i do this change? or
> is it possible to do such change?
>
> cheers,
>
> Lucas
>
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>
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