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jmatthewsr
Junior
Joined: April 09 2008
Posts: 40
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| Posted: May 01 2008 at 1:57pm | IP Logged
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Are there any known one-way audio problems in the 5.12 engine?
The following scenario is causing one-way audio:
Conference 1:
PhoneLine = 5 is connected to the conference
Local RTP port = 17408
Remote RTP port = 12874
PhoneLine = 0 is connected to the conference
Local RTP Port = 17410
Remote RTP Port = 13200
Conference 1 is successful, audio is two-way, both users hangup, the BYE/200Ok teardown is succesful.
The same 2 devices call in again and there is one-way audio:
Conference 2:
* In this conference line 4 cannot hear line 5, but line 5 can hear line 4. RTP wireshark traces confirm that no RTP audio is sent to line 4, but RTP audio is sent to line 5. RTP audio traces also show that RTP audio is received from the far end for both line 4 and line 5 and that the ip/port is correct.
PhoneLine = 4 is connected to the conference
Local RTP port = 17408
Remote RTP port = 12878
PhoneLine = 5 is connected to the conference
Local RTP Port = 17410
Remote RTP Port = 13200
Could line 5 in conference 2 still be listening on port 17408?
I can upload full traces if you think they will help.
thanks!
-justin
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support
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Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 01 2008 at 5:09pm | IP Logged
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Justin,
Are you using your own App code or one of the C++ samples that come with the media engine?
Support
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jmatthewsr
Junior
Joined: April 09 2008
Posts: 40
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| Posted: May 01 2008 at 5:40pm | IP Logged
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This was from our own app. I have the full traces that show the conference is active for each line (lanscape events). There is one-way audio so I know they are in the same conference.
I also just ran the multi-session test app , but no audio either way was heard. The UI shows that the calls entered the same confernce, but what is strange is that the URI shows <number>@255.255.255.255 for one of the calls.
Note that on this setup the conference server (either the lanscape demo or my app) is running on the same box and network card as the app server that is communicating with it. The SIP listening port is set to something other than 5060.
UAC1 (aastra) -----> App Server (public IP1):5060 ---> Conference App(public IP1):5090
UAC2 (PSTN/POTS phone)-----> PSTN Gateway ----> App Server (public IP1):5060 ----> Conference App(public IP1):5090
thanks!
-justin
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jmatthewsr
Junior
Joined: April 09 2008
Posts: 40
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| Posted: May 01 2008 at 6:03pm | IP Logged
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Another bit of information: during the one-way scenario, if I add a third participant then all 3 lines are able to hear each other and everything appears to be working (although the volume on the previously one-way path is lower than the others, this may be a phone issue).
-justin
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jmatthewsr
Junior
Joined: April 09 2008
Posts: 40
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| Posted: May 02 2008 at 12:59pm | IP Logged
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Correction to the last post. Here is a summary of the scenario I am seeing:
Caller1 dials into the conference directly over IP.
Caller2 dials into the conference via the PSTN gateway.
Caller1 can hear caller 2 but caller2 cannot hear caller1.
Caller3 dials in via the pstn gateway.
caller3 can hear caller1 but the volume is very low. Caller2 can hear caller1 and the volumesounds fine.
CAller3 hangs up and its back to the way it started, caller1 can hear caller2 but caller2 cannot hear caller1.
thanks!
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support
Administrator
Joined: January 26 2005
Location: United States
Posts: 1666
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| Posted: May 02 2008 at 1:49pm | IP Logged
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Thanks Justin,
We will make an attempt to look into this yet today.
One thing we need to do is make sure the basic conferencing logic and media exchanging is working for IP to IP calls all within your private network. In other words, does conferencing function with 2 or three IP or soft phones that reside in your network?
The last time tested the conference functionality all was OK. Possible bug we introduced? Don’t know yet.
Continue posting any info you deem helpful.
Support
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jmatthewsr
Junior
Joined: April 09 2008
Posts: 40
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| Posted: May 02 2008 at 3:36pm | IP Logged
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I noticed that the first inbound call (ip phone: UAC1) changed it's RTP source port after reINVITEing it to the conference:
UAC1 UAS1 Lanscape (conference server)
INVITE (sdp1) ->
200 (sdp2) <-
ACK->
----(RTP source: 3000)--->
INVITE (sdp1) ->
200 (sdp3) <-
ACK ->
INVITE (sdp3)<-
200 (sdp4)->
ACK <-
---(RTP source: 1000)---->
Ther is a sip-aware firewall that is changing the port in the port in sdp4, but UAS1 doesn't reINVITE the lanscape server to update it. Can you confirm that this would be a problem? The source IP is the same, but the source port changes without notification.
thanks.
-justin
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support
Administrator
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Location: United States
Posts: 1666
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| Posted: May 02 2008 at 5:15pm | IP Logged
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Justin,
A few questions before we can respond:
1) Is the UAS1 element in your picture what you are calling the “sip-aware firewall”?
2) What is the exact purpose of UAS1?
3) Does UAS1 proxy the media between UAC1 and Lanscape (conference server)?
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jmatthewsr
Junior
Joined: April 09 2008
Posts: 40
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| Posted: May 02 2008 at 5:50pm | IP Logged
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1) no, uac1 is behind a NAT: uac1->sipawarefirewall->uas1
2) b2bua (sip app server)
3) no
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support
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Location: United States
Posts: 1666
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| Posted: May 06 2008 at 3:45pm | IP Logged
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Hi Justin,
Excuse the delay in this response.
If we understand your description correctly: If UAS1 is sending a reINVITE to UAC1 and UAC1’s RTP port changes and UAC1 and “Lanscape (conference server)” exchange RTP media directly, then that will be a problem. The LanScape end of the deployment will be sending its RTP media to a destination RTP port of UAC1 that is not being used. The result will be one way audio.
Support
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jmatthewsr
Junior
Joined: April 09 2008
Posts: 40
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| Posted: May 06 2008 at 3:51pm | IP Logged
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Yep, that is my understanding as well. Thanks for the response. If the lanscape library can still detect these packets from a different source, it may be possible to have UAS1 trigger a re-invite (out of the scopt of this problem).
thanks.
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LanScape Support Forum -> LanScape VOIP Media Engine™ - Pre-Sales Technical Support
Topic: one-way audio in conference
