Thanks for the quick reply,

Here's the Wireshark capture. As you can see, we added a connection line in the second media, even though it's optional. And we don't have any video codec. I want to know if we're obliged to have at least one ?

François Roseberry

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Hi,

we've found out that our server changed the whole SDP before giving it to the callee (No matter how we changed it, it was always the same).
The SIP server we used is 3CX Phone System, from Microsoft, which I guessed is made to work with 3CX softphone, so the replaced SDP does the work for this. We found another free and more "transparent" SIP server, which ias a lot easier, more user-friendly, it's called minisipserver.

Now the SDP passes, and the two media lines are established, and we can talk and hear each other through the audio line, just as normal.
But we added a dummy video codec (which doesn't exist), rtpmap:2 BITMAP/16000, before the telephone-events line. Otherwise, it crashed at the media session creation. But if you say that telephony-events attribute is not mandatory, we'll try to remove both.

We understand what the attributes connection, rtcp and sendrecv (media direction) are for, but what about fmtp and rtpmap ?
We have a vague idea, since the rtpmap attribute of the already created audio "m" line contains info about a codec, like PCMU or PCMA, and a supported bitrate. There's also the telephony-events that we're not sure about.

If you could just give us a little hint on that,

and we'll keep up,
thanks

François Roseberry

Date: Wed, 25 Mar 2009 15:40:24 +0000
From: bennylp@teluu.com
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] SDP Question

2009/3/25 François Roseberry frank_pharaoh05@hotmail.com

Thanks for the quick reply,

Here's the Wireshark capture. As you can see, we added a connection line in the second media, even though it's optional.
That should be okay, it will be picked up if it's present.

And we don't have any video codec. I want to know if we're obliged to have at least one ?

That's fine too. You can also omit the telephone-event and just put empty format list in the m=video line, which is valid according to the spec.

At glance, the SDP looks okay, I couldn't find anything wrong with it either. Perhaps the best way to troubleshoot this is to put breakpoint on pjmedia_sdp_parse() function in pjmedia/sdp.c, and trace it step by step to know what the parser thinks about it. I'll do that as soon as time permits.

cheers
Benny

François Roseberry

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Now the moment to send data on the media port (by using pjmedia_port_put_frame()) an it crashes.
We looked at it and we found out that it asks for the stream codec to encode the buffer before sending it out. Since we have no codec, i guess we'll have to code one. But we don't want it complicated for the beginning, we just want a plain codec that does nothing except declaring the appropriate methods. Can you give us a guideline on how to develop our dummy codec and integrate it in PJSIP (I haven't found any tutorial on that) ?

Or maybe there's an other way that we haven't thought of ?

François Roseberry

Date: Thu, 26 Mar 2009 06:49:06 +0000
From: bennylp@teluu.com
To: pjsip@lists.pjsip.org
Subject: Re: [pjsip] SDP Question

2009/3/25 François Roseberry frank_pharaoh05@hotmail.com

Hi,

we've found out that our server changed the whole SDP before giving it to the callee (No matter how we changed it, it was always the same).
The SIP server we used is 3CX Phone System, from Microsoft, which I guessed is made to work with 3CX softphone, so the replaced SDP does the work for this. We found another free and more "transparent" SIP server, which ias a lot easier, more user-friendly, it's called minisipserver.

That explains it!
Now the SDP passes, and the two media lines are established, and we can talk and hear each other through the audio line, just as normal.

But we added a dummy video codec (which doesn't exist), rtpmap:2 BITMAP/16000, before the telephone-events line. Otherwise, it crashed at the media session creation.
Is that in pjmedia_session_create()? Is it crash or assertion?

But if you say that telephony-events attribute is not mandatory, we'll try to remove both.

I don't think telephony-events is right for m=video (it doesn't sound like making sense).

Sorry I think I was wrong about empty format list in SDP m= line. Syntactically, the spec does allow empty format list, but in pjsip we only accept that if port number is zero. So you need to codec indeed.

We understand what the attributes connection, rtcp and sendrecv (media direction) are for, but what about fmtp and rtpmap ?

rtpmap is mandatory for dynamic PT. fmtp is additional info about the codec, and it's codec specific. Its use should be described in the corresponding audio/video profile RFC I think.

-benny

We have a vague idea, since the rtpmap attribute of the already created audio "m" line contains info about a codec, like PCMU or PCMA, and a supported bitrate. There's also the telephony-events that we're not sure about.

If you could just give us a little hint on that,

and we'll keep up,
thanks

François Roseberry

Date: Wed, 25 Mar 2009 15:40:24 +0000
From: bennylp@teluu.com
To: pjsip@lists.pjsip.org

Subject: Re: [pjsip] SDP Question

2009/3/25 François Roseberry frank_pharaoh05@hotmail.com

Thanks for the quick reply,

Here's the Wireshark capture. As you can see, we added a connection line in the second media, even though it's optional.
That should be okay, it will be picked up if it's present.

And we don't have any video codec. I want to know if we're obliged to have at least one ?

That's fine too. You can also omit the telephone-event and just put empty format list in the m=video line, which is valid according to the spec.

At glance, the SDP looks okay, I couldn't find anything wrong with it either. Perhaps the best way to troubleshoot this is to put breakpoint on pjmedia_sdp_parse() function in pjmedia/sdp.c, and trace it step by step to know what the parser thinks about it. I'll do that as soon as time permits.

cheers
Benny

François Roseberry

Messenger vous offre des tonnes de nouvelles fonctions qui rendent le clavardage encore plus plaisant Cliquez ici pour en savoir plus.

Visit our blog: http://blog.pjsip.org

pjsip mailing list

pjsip@lists.pjsip.org

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