| Summary: | ASTERISK-00470: RTP problems with Cisco IP Phones (SIP) | ||
| Reporter: | lele (lele) | Labels: | |
| Date Opened: | 2003-10-31 13:52:52.000-0600 | Date Closed: | 2011-06-07 14:05:29 |
| Priority: | Major | Regression? | No |
| Status: | Closed/Complete | Components: | Core/General |
| Versions: | Frequency of Occurrence | ||
| Related Issues: | |||
| Environment: | Attachments: | ||
| 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. | ||