|Summary:||ASTERISK-00470: RTP problems with Cisco IP Phones (SIP)|
|Date Opened:||2003-10-31 13:52:52.000-0600||Date Closed:||2011-06-07 14:05:29|
|Description:||When using Cisco SIP IP Phones (both 7960 and 7905) with asterisk on average latency networks ( ~ 50ms, weirdly it doesn't happen on LAN), about two seconds of audio from * to phone can be heard, after that only silence with few sparse frames. It happens with any codec and regardless of call destination (ie: prompts, bridged calls). Other direction (from phone to * is unaffected).|
On the phone diagnostic screen it shows the jitter increasing to unrealistic high levels and the phone starts dropping frames after the first 100/150.
Putting the call on hold and restoring it from the hold status restores a perfect audio which remains stable indefinitely.
I would suspect some timestamping issue within the first few RTP frames. Other clients (pingtel, X-ten) do not have the symptom.
|Comments:||By: Brian West (bkw918) 2003-10-31 17:42:57.000-0600|
what firmware are you running? Because my 7960 doesn't do what you describe. can you include your sip.conf entry?
By: Brian West (bkw918) 2003-10-31 17:43:37.000-0600
Oh can you include your cnf file for inspection?
By: lele (lele) 2003-10-31 21:07:39.000-0600
I found the problem and doesn't have anything to do with asterisk. It was a bug in rtp header compression in both the routers I used for testing that corrupted header data. Weird it didn't affect pingtels and X-ten.
edited on: 10-31-03 21:06
By: John Todd (jtodd) 2003-11-02 13:53:55.000-0600
So can we close this out? I'm going to assume "yes" and we can re-open if required.