RTP issues can be difficult to diagnose without a healthy amount of packet captures and other troubleshooting information.
This document was written in 2008 and is likely no longer applicable.
Devices with Issues
Carriers known to be using this device: L3, WilTel, Bandwidth.com, Broadvox, Global Crossing, XO, Qwest and T-Systems (a division of Deutsche Telekom).
- 19:40, 9 December 2008 (PST)
- Known to support exactly G.729 with rfc2833 OR G.711u with DTMF inband. If you're using FreeSWITCH then you can purchased a license for G.729 or use G.711u (PCMU) with DTMF inband. ( also RFC2833 look below how )
- Dropped audio infrequently. Anthony discovered that they have a +500ms response time if you STOP streaming RTP data to the carrier.
Whatever hardware they use has a list of reasons to reset.
- If your installation of FreeSWITCH wants to change timestamp base and send them mark bit, they reset with 2 seconds of silence.
- If you send RFC2833 on it's own timestamp base it resets.
If you are suffering from dropped audio, specifically regarding the first 2-3 seconds of a single audio stream then try the following:
Take care when using this parameter, as it may suppress ringback tones by actually sending early media.
If you are having DTMF problems and Sonus is in your media path, you should make sure you are using the latest version of FreeSWITCH. As of FreeSWITCH revision 10744, FreeSWITCH auto-detects Sonus end-points and applies a hack to "fix" the issue. The hack is described as follows (from switch_types.h):
Sonus wrongly expects that, when sending a multi-packet 2833 DTMF event, the sender should increment the RTP timestamp in each packet when, in reality, the sender should send the same exact timestamp and increment the duration field in the 2833 payload. This allows a reconstruction of the duration if any of the packets are lost. final_duration - initial_timestamp = total_samples However, if the duration value exceeds the space allocated (16 bits), the sender should increment the timestamp one unit and reset the duration to 0. Always sending a duration of 0 with a new timestamp should be tolerated but is rarely intentional and is mistakenly done by many devices. The issue is that the Sonus expects everyone to do it this way instead of tolerating either way. Sonus will actually ignore every packet with the same timestamp before concluding if it's DTMF.
Note that the version of FreeSWITCH which has this patch auto-detects Sonus, so you don't need to configure anything.
If you are going through a provider who uses Sonus only as an SBC and not in your media path FreeSWITCH may incorrectly identify your call as going through Sonus and actually corrupt your media stream's DTMF. You can override FreeSWITCH's auto-detection of end-user agents with a flag in profiles stored under $FS_ROOT/sip_profiles/ . Just add:
In some cases, you may have the reverse situation where Sonus or Cisco equipment sits in your media path, but FreeSWITCH can't auto-identify it. In that case, try these settings:
If your call to a Sonus still won't work
For those of you who have tried the auto-rtp-bugs and still cannot get DTMF to work for G711 negotiated calls, there appears to be an issue with the Sonus using Fax treatment which drops the out of band DTMF packets that come from FreeSWITCH (and other platforms). Some more in-depth information for Sonus users (at least for 5.0 and 5.1 users, I can't speak for those who may have the latest and greatest code revisions which might fix this) is below.
In the GSX Navigator (Sonus Insight), go to Sonus→Profiles→Packet Service Profiles→default. You may have some changes made to the Fax Treatment/Failure section. To resolve the DTMF you will have to change that to match the below examples.
An example of a base profile that will not work:
Audio Encodings: G.711 DTMF Modem Fax Pkt Size/ Law/ Relay/ Treatment/ Treatment/ Priority Codec Code Rate Send SID Remove Failure Failure 3 G711SS 20 LAWFROMCIR NONE G711FALLBACK G711FALLBACK 0BPS DISABLED DISABLED CONTINUE CONTINUE 4 G711 20 LAWFROMCIR NONE G711FALLBACK G711FALLBACK 0BPS DISABLED DISABLED CONTINUE CONTINUE
What you would need to change it to, to fix the DTMF:
Audio Encodings: G.711 DTMF Modem Fax Pkt Size/ Law/ Relay/ Treatment/ Treatment/ Priority Codec Code Rate Send SID Remove Failure Failure 3 G711SS 20 LAWFROMCIR NONE G711FALLBACK FAXRELAYORG711FA 0BPS DISABLED DISABLED CONTINUE CONTINUE 4 G711 20 LAWFROMCIR NONE G711FALLBACK FAXRELAYORG711FA 0BPS DISABLED DISABLED CONTINUE CONTINUE
The above profile will work if you are sending direct to the GSX using the dtg=TRUNKGROUP protocol. If you are routing your calls to the PSX, then you would make the appropriate changes to the Preferred Packet Service Profile that is attached to the trunk group that you are a) sending from or b) sending to.
Voice Operator Panel (VOP)
VOP-Softclients don't like changes in the RTP timestamp, e.g. when a REFER to the VOP-client occurs and the RTP-stream is replaced by the stream from a different call with a different timestamp. This leads to one-way audio for the callee, the VOP Softphone doesn't process the incoming RTP packets with the "new" timestamp and the callee will not hear the caller.
Setting rtp-rewrite-timestamps to True in the affected SIP-profile eventually solves the problem.
For more information on these and other Sonus issues: