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how CSS will treat incoming RTSP request?

nayanpanchal
Level 1
Level 1

We are introducing one more service to serve live streaming traffic. I have prepared a configuration for the same too. My worry is that incoing url will be " rtsp://liv.tataindicom.com:554". By default, CSS will consider it as a http traffic. If I configure, CSS in the following way, will it work?

content RTSP

vip address 172.23.2X2.XX

protocol tcp

port 554

add service RTSP_Live

application realaudio-control

flow-timeout-multiplier 10

url "//liv.tataindicom.com:8080/*"

active

content RTSP-UDP

vip address 172.23.2X2.XX

protocol udp

url "//liv.tataindicom.com:8080/*"

flow-timeout-multiplier 10

active

Thanks in advance

8 Replies 8

nayanpanchal
Level 1
Level 1

okey friends,

i tried this and it failed misrebly. I have following observations:

1.even if i add url "/*" in content RTSP, it stops working.

2. somehow i have this feeling that RTSP and url need some special commands.

Here we are using two types of traffic (Both RTSP) with same port, protocol, VIP but different url.

I have raised a TAC too n both of us are struggling bad.

Any input would be appreciated.

Best Regards

first, the url command can only be used for HTTP traffic - the css will be looking for "http://"

So, you can't use it for rtsp.

Same for udp traffic, the command can't be used on such content rule.

Also, the 'application real audio' is really not going to do what you expect.

This command was introduced a long time ago for a specific issue.

The CSS actually does not understand RTSP protocol.

So, if you can make this work at level 3 or 4 [ip,tcp,udp] then that's fine.

Just be aware that since the CSS does not understand rtsp traffic it won't be able to change the payload andyou might see the client trying to connect directly to the server instead of the vip.

I would recommend not to loadbalance rtsp with the CSS.

If you need to loadbalance this protocol, the CSM or the ACE module would be a better option.

Gilles.

Dear Gilles,

You are perfectly right. I guess I assumed too much with regard to URL command under content rule. TAC told me that url will work ONLY for http protocol.

We may have to come up with different port number as different VIP is not possible in our case.

By the way, does realaudio-control has anything to do with "two minute disconnect" problem?

We are using software version 7.2 (build 206).

Thanks in advance.

disconnect is often the result of idle session.

The CSS has a very short idle timeout by default.

You may want to try to use the command 'flow-timeout-multiplier' and see if this improves the 2 min disconnect.

If not at all, a sniffer trace might be needed to verify what is happening exactly.

Finally, I would definitely recommend an upgrade to 7.50 or 8.10. We do not do software fix for 7.20. So in case of issue, you will be asked an upgrade first.

Gilles.

Hey Gilles,

Thanks for your input, Two things in this reply.

When we faced this "2 minute diconnect" issue first time, yes flow-timeout-multiplier did resolve the problem. we are using value of 10.

We are load balancing RTSP using Layer 4 rule. It is working fine but we cannot introduce another RTSP service using same L3, L4 parameters. But here, clients (Mobile Handsets) are the culprits.

We are using two content rule, one using TCP (for session establishment, port 554) and second using UDP (data channel).

Thanks for your suggetion of upgradation. We definately need to give that a thaught.

thanks,

nayan

Hi Gilles ,

what is happen , when i use protcol any instead of TCP or UDP in the Content ???

Can the CSS work then with RTSP on mobile phones ??

Thx in advance for an answer.

bye joerg

Joerg,

I didn't see in the previous discussion what was the problem - no sniffer trace.

Therefore, I can't tell if a proposed solution would fix something or not.

First step would be to make sure we know the cause of the problem - not the symptoms.

Anyway, changing the rule from L4 to L3 would have no impact. They are processed exactly the same way internally.

Gilles.

Hi Gilles,

i have in the moment problems also with RTSP, we have moved from an Alteon Switch to an CSS some SMS services and now this is not working anymore with the RTSP.I have only configured TCP + Port Number, not more. When now the Server sends an UDP data Stream what is happen on the CSS ?

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