[SIPForum-discussion] Unable to receive RTP Stream over NAT
Anuroop
anuroop_r at dataone.in
Thu Apr 5 14:58:03 UTC 2007
Hello Anil.
--> What type of NAT are you benind? It is is symmetric NAT, STUN may
not be sufficient.
--> For every RTP port to receive data, there should be an RTCP port for
receiving control packets.
--> You need to create pin holes for the media to get pass through once
ports are identified.
Regards,
Anuroop.
-----Original Message-----
From: discussion-bounces at sipforum.org
[mailto:discussion-bounces at sipforum.org]On Behalf Of anilkumar mantena
Sent: Thursday, April 05, 2007 8:09 PM
To: discussion at sipforum.org
Subject: [SIPForum-discussion] Unable to receive RTP Stream over NAT
Hi all,
I tried to implement a program for RTP Streaming over Internet (VOIP). I
am able to register with the proxy and establishing a session with INVITE.
But I'm not able to receive the voice and even I am not able to send voice.
I used STUN (stun4j) to detect my public ip and public port using
addressDiscovery.determineAddress() method. I tried by using private ip and
port; public ip and public port (found in STUN Discovery); and used Media IP
and Media Port (from SDP) to listen/send RTP stream over the Internet.
Still no use of it. I'm unable to send/receive voice.
1. What might be the problem?
2. What are the ports I need to configure for Listening and for sending?
3. When I should do STUN Discovery?
4. Is Discovery process is sufficient or shall I do any binding (if yes
please mention the methods in stun4j)
5. Any other solutions to solve this problem?
Thanks in advance.....
Anilkumar Manthena
----------------------------------------------------------------------------
--
It's here! Your new message!
Get new email alerts with the free Yahoo! Toolbar.
-------------- next part --------------
An HTML attachment was scrubbed...
URL: <http://sipforum.org/pipermail/discussion/attachments/20070405/7a23bbbd/attachment-0002.html>
More information about the discussion
mailing list