I am trying to get Freeswitch to work with my SIP provider, IP24, but I always get very bad sound in one direction. I made a wireshark trace, and looked at the jitter graph it produced. I made an incoming call from my mobile phone to Freeswitch via SIP provider IP24, PCMA codec: 1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from freeswitch->ip24 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24->freeswitch There is a big difference in jitter between the two graphs, and the speech quality is very bad ip24->freeswitch (2). What can I do to fix the speech quality? Asterisk and OpenPBX (CallWeaver) have both excellent audio quality in calls through my SIP provider IP24. Here is a voip graph of the SIP call: http://www.myrvold.org/ freeswitch/graph.jpg Ivan
Oops, fixed the graph: http://www.myrvold.org/freeswitch/graph.jpg Ivan Den 6. mai. 2007 kl. 13:11 skrev Ivan C Myrvold:
I am trying to get Freeswitch to work with my SIP provider, IP24, but I always get very bad sound in one direction. I made a wireshark trace, and looked at the jitter graph it produced.
I made an incoming call from my mobile phone to Freeswitch via SIP provider IP24, PCMA codec:
ip24
1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from freeswitch- 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24-
freeswitch
There is a big difference in jitter between the two graphs, and the speech quality is very bad ip24->freeswitch (2).
What can I do to fix the speech quality? Asterisk and OpenPBX (CallWeaver) have both excellent audio quality in calls through my SIP provider IP24.
Here is a voip graph of the SIP call: http://www.myrvold.org/ freeswitch/graph.jpg
Ivan
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Did you happen to update to the latest code that includes a jitterbuffer as of friday? /b On May 6, 2007, at 6:30 AM, Ivan C Myrvold wrote:
Oops, fixed the graph:
http://www.myrvold.org/freeswitch/graph.jpg
Ivan
Den 6. mai. 2007 kl. 13:11 skrev Ivan C Myrvold:
I am trying to get Freeswitch to work with my SIP provider, IP24, but I always get very bad sound in one direction. I made a wireshark trace, and looked at the jitter graph it produced.
I made an incoming call from my mobile phone to Freeswitch via SIP provider IP24, PCMA codec:
ip24
1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from freeswitch- 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24-
freeswitch
There is a big difference in jitter between the two graphs, and the speech quality is very bad ip24->freeswitch (2).
What can I do to fix the speech quality? Asterisk and OpenPBX (CallWeaver) have both excellent audio quality in calls through my SIP provider IP24.
Here is a voip graph of the SIP call: http://www.myrvold.org/ freeswitch/graph.jpg
Ivan
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I used the svn trunk from yesterday. When I start it, it says: FreeSWITCH Version 1.0.5093 Started. Ivan Den 6. mai. 2007 kl. 18:30 skrev Brian West:
Did you happen to update to the latest code that includes a jitterbuffer as of friday?
/b
On May 6, 2007, at 6:30 AM, Ivan C Myrvold wrote:
Oops, fixed the graph:
http://www.myrvold.org/freeswitch/graph.jpg
Ivan
Den 6. mai. 2007 kl. 13:11 skrev Ivan C Myrvold:
I am trying to get Freeswitch to work with my SIP provider, IP24, but I always get very bad sound in one direction. I made a wireshark trace, and looked at the jitter graph it produced.
I made an incoming call from my mobile phone to Freeswitch via SIP provider IP24, PCMA codec:
ip24
1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from freeswitch- 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24-
freeswitch
There is a big difference in jitter between the two graphs, and the speech quality is very bad ip24->freeswitch (2).
What can I do to fix the speech quality? Asterisk and OpenPBX (CallWeaver) have both excellent audio quality in calls through my SIP provider IP24.
Here is a voip graph of the SIP call: http://www.myrvold.org/ freeswitch/graph.jpg
Ivan
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It turned out that SIPURA SPA-2000 has a bug (which also affects PAP2). But easily fixed: Use your browser to open the SIP configuration on the SIPURA SPA-2000. Change "RTP Packet Size" from 0.030 to the new value 0.020 Now I have perfect audio both ways with Freeswitch. Why didn't I have the problem with Asterisk? Apparently Asterisk ignores the value set by the SIPURA. Ivan Den 6. mai. 2007 kl. 13:11 skrev Ivan C Myrvold:
I am trying to get Freeswitch to work with my SIP provider, IP24, but I always get very bad sound in one direction. I made a wireshark trace, and looked at the jitter graph it produced.
I made an incoming call from my mobile phone to Freeswitch via SIP provider IP24, PCMA codec:
ip24
1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from freeswitch- 2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24-
freeswitch
There is a big difference in jitter between the two graphs, and the speech quality is very bad ip24->freeswitch (2).
What can I do to fix the speech quality? Asterisk and OpenPBX (CallWeaver) have both excellent audio quality in calls through my SIP provider IP24.
Here is a voip graph of the SIP call: http://www.myrvold.org/ freeswitch/graph.jpg
Ivan
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I don't think this value can be ignored, it's just adopted to that value. G729 should be 20 by standard... What codec did you use? On 7 May 2007 21:09, Ivan C Myrvold wrote:
It turned out that SIPURA SPA-2000 has a bug (which also affects PAP2). But easily fixed:
Use your browser to open the SIP configuration on the SIPURA SPA-2000. Change "RTP Packet Size" from 0.030 to the new value 0.020
Now I have perfect audio both ways with Freeswitch.
Why didn't I have the problem with Asterisk? Apparently Asterisk ignores the value set by the SIPURA.
Ivan
Den 6. mai. 2007 kl. 13:11 skrev Ivan C Myrvold:
I am trying to get Freeswitch to work with my SIP provider, IP24, but I always get very bad sound in one direction. I made a wireshark trace, and looked at the jitter graph it produced.
I made an incoming call from my mobile phone to Freeswitch via SIP provider IP24, PCMA codec:
1. http://www.myrvold.org/freeswitch/me2ip24.jpg , from freeswitch-
ip24
2. http://www.myrvold.org/freeswitch/ip242me.jpg , from ip24-
freeswitch
There is a big difference in jitter between the two graphs, and the speech quality is very bad ip24->freeswitch (2).
What can I do to fix the speech quality? Asterisk and OpenPBX (CallWeaver) have both excellent audio quality in calls through my SIP provider IP24.
Here is a voip graph of the SIP call: http://www.myrvold.org/ freeswitch/graph.jpg
Ivan
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There were some bugs on the sipura devices where it would properly negotiate 30 ms, but if you actually sent it 30 ms packets, the audio quality was horrible as it was still expecting 20ms packets. The only way to fix it is to change that setting. Mike
-----Original Message----- From: freeswitch-users-bounces@lists.freeswitch.org [mailto:freeswitch- users-bounces@lists.freeswitch.org] On Behalf Of Vazir Sent: Tuesday, May 08, 2007 12:31 AM To: freeswitch-users@lists.freeswitch.org Subject: Re: [Freeswitch-users] Solved: Jitter problems
I don't think this value can be ignored, it's just adopted to that value. G729 should be 20 by standard... What codec did you use?
participants (4)
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Brian West
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Ivan C Myrvold
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Michael Jerris
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Vazir