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Automatically exported from code.google.com/p/sipme-media-server
It's an used accessor at this point, but might as well correct it.
/**
* Gets the upper boundary of available range.
* @retun min port number
*/
public int getHighestPort() {
return portManager.getLowestPort();
}
Original issue reported on code.google.com by [email protected]
on 23 Aug 2013 at 4:22
Add announcement support to custom endpoints imlemented for sbc.
this may be complicated since mixing is not supported in this endpoints and not
always mms will do transcoding ( it may use codec negotiated between 2 sides
even without having it implemented ) , therefore it will not always be possible
to transcode audio to correct destination codec
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:38
Add H.248 Support for Media Server
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 5:07
Add H.263+ Codec Support
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:45
Media server could not handle with Media Format:97 telephone-event. When
incoming SDP Offer contains rtpmap:97 telephone-event/8000, the replied
SDP Answer doesn't contains this Media Attribute type. Then cause fail to
detect DTMF on that call session.
To solve this issue
Yulian Oifa suggest making the following change:
In
https://code.google.com/p/sipme-media-server/source/browse/io/rtp/src/main/java/
ua/mobius/media/server/impl/rtp/sdp/RTPFormats.java#192
Change from
for (int i = 0; i < this.rtpFormats.size(); i++) {
for (int j = 0; j < other.size(); j++) {
if (this.rtpFormats.get(i).getFormat().matches(other.rtpFormats.get(j).getFormat())) {
if(this.rtpFormats.get(i).getFormat().getName().equals(AVProfile.telephoneEvent.getName()))
res.add(this.rtpFormats.get(i));
else if(!hasNonDtmf)
{
res.add(this.rtpFormats.get(i));
hasNonDtmf=true;
}
}
}
}
To:
for (int i = 0; i < this.rtpFormats.size(); i++) {
if(this.rtpFormats.get(i).getFormat().getName().equals(AVProfile.telephoneEvent.getName()))
res.add(this.rtpFormats.get(i));
else if(!hasNonDtmf)
{
for (int j = 0; j < other.size() && !hasNonDtmf; j++) {
if (this.rtpFormats.get(i).getFormat().matches(other.rtpFormats.get(j).getFormat()))
{
res.add(this.rtpFormats.get(i));
hasNonDtmf=true;
}
}
}
}
Original issue reported on code.google.com by [email protected]
on 13 Oct 2014 at 1:55
GSM EFR implementation for Media Server
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:26
Currently implemented over mgcp and uses threads and runnable in non optimal
way.
Need to add built in support for JSR in media server and reimplement driver
based on new support.
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 5:06
More Details :
https://groups.google.com/forum/#!topic/sipme-media-server/J7WOR0aArxE
Original issue reported on code.google.com by [email protected]
on 23 Jan 2014 at 12:26
In this mode sbc should set to top of list codecs received from first UA, but
will add supported under transcoding codecs after that.Depending on answer
received from second UA endpoint will work either as pass through ( if both
sides support any of codecs ) or transcoded otherwise.
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:36
Test maximum load for OPUS codec.
Set target and optimized implementation
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:27
When other side does not supports OOB currently ignores and does not sends data
, should translate it to inband and send
Original issue reported on code.google.com by [email protected]
on 4 Aug 2013 at 10:53
Add H.264 Video codec support.
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:39
Test maximul load for GSM + GSM EFR codecs.
Set target and optimize both codecs
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:28
Add RTCP mux support for Media Server
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:58
Add RTCP support for Media Server
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:55
Add Standalone ( Independent ) centralized management interface that will allow
to manager all the server for specific company.
Implement web interface , add web services support to allow querying data for
other apps that may be used already.
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 5:03
See thread:
https://groups.google.com/forum/#!topic/sipme-media-server/9nv2WGDbY6A
Original issue reported on code.google.com by [email protected]
on 19 Mar 2015 at 12:38
Implement driver for db access to obtain MMS configuration , logging storing ,
etc.
At first step use cassandra NOSQL DB.Other may be added in future depending on
community interest
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 5:02
Add SNMP monitoring support.
Add endpoint , resources and connections params
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:59
1. Bugs exist in
org.mobicents.javax.media.mscontrol.mediagroup.MediaGroupImpl#nextRequestID
2. Constructor of jain.protocol.ip.mgcp.message.parms.RequestIdentifier is
expecting unsigned hexadecimal string.
3. Exception thrown when
org.mobicents.javax.media.mscontrol.MediaSessionImpl#getUniqueReqID reaches
80000000,
java.lang.IllegalArgumentException: Request identifier must contain between 1 and 32 hexadecimal digits!
Code to quick test:
public static void main(String args[]){
int reqId = 0;
while(true){
try {
new RequestIdentifier(Integer.toString(++reqId));
} catch (IllegalArgumentException ex){
ex.printStackTrace();
break;
}
}
}
What is the expected output? What do you see instead?
org.mobicents.javax.media.mscontrol.mediagroup.MediaGroupImpl#nextRequestID
should not throw exception when integer overflow.
What version of the product are you using? On what operating system?
Please provide any additional information below.
Original issue reported on code.google.com by [email protected]
on 28 May 2014 at 1:39
Attachments:
Currently mms runs stable therefore logging of datetime is not enough to get
more details for specific event.
Need to change logging format to include date also
Original issue reported on code.google.com by [email protected]
on 23 Jan 2014 at 12:27
Test maximum load for g729 codec.
Set target and optimize codec to reach the target
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:30
Add Fax Support to Media Server
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:54
Allow Connection state timeouts configuration in management interface.
Currently are static / non configurable
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 5:04
Need to add adaptive jitter buffer support for mms.
Currently it has static size and does not takes into account client network
which may produce delay / jitter much higher then 60ms and therefore cause bad
audio quality.
Original issue reported on code.google.com by [email protected]
on 25 Nov 2013 at 7:51
Add RTP/AVPF Support For Media Server
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:56
Add Opus support to media server.
Planned to be added both native ( webrtc based ) and java based implementations.
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:26
Add support for Secure Real-time Transport Protocol (SRTP) / (IANA SDP code
RTP/SAVP) for Media Server
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:58
Add Video Player And Video Recording implementation
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:50
Test maximum load for ILBC codec.
Set target and optimize codec.
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:30
Add compilation for native ilbc + native opus for windows
Modify bootstrap pom to copy files automatically to native folder even when
file does not exists.
Original issue reported on code.google.com by [email protected]
on 28 Oct 2013 at 6:20
Add custom endpoint implementation which will negotiate sdp based on UA's sdp
received ( thus codecs received from first UA connection will be duplicated to
second one ).
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:33
Add Secure RTP Support for (RTCP)-Based Feedback (RTP/SAVPF)
Original issue reported on code.google.com by [email protected]
on 20 Jul 2013 at 4:57
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