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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