RTP Wrong Timestamps and audio hiccups inbound audio

[Murray Leach]

Has anyone experienced an odd issue with seemingly random call quality on the inbound audio on lots of calls?

I get some missing audio which is not related to lost packets or jitter.

2 sip phones and a soft phone tested on 3 different networks all claim neither is present.

I wiresharked the calls and played back the audio - sounds exactly as it was but wireshark shows a yellow W at every moment of interrupted audio.

Apparently its a wrong timestamp... If you play the audio while ignoring the timestamps then its crystal clear and as expected.

Where would a bad timestamp come from and how would I fix it?

Thanks guys

[m.l...@mergedcomms.com]

I have been pointed in the direction of http://wiki.freeswitch.org/wiki/Sofia.conf.xml#rtp-rewrite-timestamps

Any advice?

[Arne van Balgoijen]

Murray,

We had similar issues and changed this setting 

<param name="rtp-rewrite-timestamps" value="true"/>

in the file conf/sip_profiles/external.xml

One of our trunk providers even dropped calls before the change, definitely give it a try.

Br Arne

[m.l...@mergedcomms.com]

This seems to have made a great improvement but not totally solved the issue - perhaps its our hosting environment - something has changed?

So thanks for making it useable at least

[Darren McIntyre]

Has anyone figured this problem out? I have having the exact same problem with my many of my VOIP calls.  The problem has just recently "started", but has been going on for about 3 months and I have not been able to isolate the issue.  Any suggestions, feedback, or guidance would be appreciated. 

[Darren Schreiber]

Audio is 100% FreeSWITCH handled in terms of quality and transcoding and such. You might want to post this question to #freeswitch mailing list

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[Arne van Balgoijen]

Did you try the setting

<param name="rtp-rewrite-timestamps" value="true"/>

in Freeswitch config 

conf/sip_profiles/external.xml

It fixed a lot of similar problems for us.

[VCCS Telecom]

Can you be more specific about network architecture? Are the clients behind NAT and have you tested calls to a static endpoint? I would do this first. Many end users require certain firewall rules for rtp traffic etc...this is way too commonly the issue.

Secondly, rewrite wont fix the whole issue if packets are getting dropped along the way even though you suspect they are not. Of course due to the fact that udp doesnt have error correction...