And FTR, this *is* on the wiki - I just didn't know about it. :)<div><br></div><div><a href="http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation">http://wiki.freeswitch.org/wiki/Channel_Variables#SDP_Manipulation</a></div>
<div>(Last item)</div><div><br></div><div>-MC<br><br><div class="gmail_quote">On Wed, Jun 15, 2011 at 3:47 PM, Anthony Minessale <span dir="ltr"><<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex;">However, there is not really any rule anywhere that states that bypass<br>
media will not touch the packets.<br>
All of the packets/sdp are still subject to SIP SOA model so if you<br>
pass a remote sdp into the lib, sofia will parse and modify it.<br>
The idea of bypass media is to bypass the media by advertising the<br>
proper codecs/addrs to arrange point to point media and there is no<br>
assurance that the packets will not be modified.<br>
<br>
However,<br>
<br>
There is a sofia param called "enable-soa" and a variable<br>
sip_enable_soa for per call basis which can be set to false to disable<br>
SIP SOA from sofia and most likely result in untouched exchange of<br>
SDP.<br>
<br>
try this in your dialplan<br>
<br>
<action application="set" data="bypass_media=true"/><br>
<action application="export" data="sip_enable_soa=false"/><br>
<div><div></div><div class="h5"><br>
<br>
<br>
<br>
On Wed, Jun 15, 2011 at 5:11 PM, Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>> wrote:<br>
> Ah, quite right. I'd file this in Jira. Hopefully it's not buried deep in<br>
> the Sofia stack...<br>
> -MC<br>
><br>
> On Wed, Jun 15, 2011 at 2:58 PM, Kristian Kielhofner <<a href="mailto:kris@kriskinc.com">kris@kriskinc.com</a>><br>
> wrote:<br>
>><br>
>> Michael,<br>
>><br>
>> This isn't an SDP offer/answer issue with FreeSWITCH. Even if there<br>
>> were underlying offer/answer issues between the endpoints FreeSWITCH<br>
>> is in bypass_media mode and it shouldn't care about the SDP because by<br>
>> definition (bypass_media) the SDP is "pass through" between the two<br>
>> remote endpoints.<br>
>><br>
>> The specific issue lies in the 183 w/ SDP at line 207. It has<br>
>> rfc2833 payload type 110. When FreeSWITCH forwards this to the other<br>
>> end (line 276) the SDP is untouched (expected with bypass_media)<br>
>> EXCEPT the rfc2833 payload type has been changed to 101 (unexpected<br>
>> with bypass_media). It does this again with the 200 OK. All other<br>
>> SDP params are passed through - media address, port, codec, goofy<br>
>> session name, and even the SER rtpproxy attributes. I'll admit the<br>
>> remote end/proxy is doing some strange stuff - 100, 101, 183, then<br>
>> 180, etc but this does look like a strange bypass_media bug in<br>
>> FreeSWITCH. The RFC2833 payload type should be forwarded between the<br>
>> two remote endpoints without being modified by FreeSWITCH - just like<br>
>> all of the other SDP parameters (or any part of the SIP body, for that<br>
>> matter).<br>
>><br>
>> On Wed, Jun 15, 2011 at 5:09 PM, Michael Collins <<a href="mailto:msc@freeswitch.org">msc@freeswitch.org</a>><br>
>> wrote:<br>
>> > QQ,<br>
>> > I don't see that FreeSWITCH is doing anything incorrect here. According<br>
>> > to<br>
>> > RFC3264, the offerer (FreeSWITCH) sends an SDP and in the case of RTP,<br>
>> > the<br>
>> > answerer (GRSIP Gateway) must use the payload type offered, even if the<br>
>> > answerer uses a different payload type when it sends a telephone-event.<br>
>> > <a href="http://tools.ietf.org/html/rfc3264#section-6.1" target="_blank">http://tools.ietf.org/html/rfc3264#section-6.1</a><br>
>> > Specifically near the end:<br>
>> > "In the case of RTP, it MUST use the payload type numbers<br>
>> ><br>
>> > from the offer, even if they differ from those in the answer."<br>
>> ><br>
>> > Technically, FreeSWITCH isn't "changing" anything anyway. The originator<br>
>> > of<br>
>> > the call (Vox Callcontrol) is the one who chose PT 101 in the INVITE<br>
>> > that it<br>
>> > sent to FreeSWITCH. FS is just passing that along without modifying it.<br>
>> > I think you need to contact the people running the gateway and make sure<br>
>> > that they understand that they are not following RFC3264 if they're<br>
>> > rejecting telephone-events in PT 101 simply because they prefer to send<br>
>> > in<br>
>> > PT 110. Also, if your Vox Callcontrol client has any configuration<br>
>> > options<br>
>> > then maybe you can tell it to use PT 110 for RFC2833 DTMFs. It is a bit<br>
>> > of a<br>
>> > workaround but this is SIP, so no one is expecting perfection. :)<br>
>> > -MC<br>
>><br>
>> --<br>
>> Kristian Kielhofner<br>
>><br>
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<br>
</div></div>--<br>
Anthony Minessale II<br>
<br>
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