I need to do a bit more work on this before I have enough information to submit a Jira but just on the off-chance anyone else has had this issue: We have handsets using GSM with 120ms ptime but when this gets transcoded to something else (for example 20ms PCMU) the resulting audio is choppy as though there was a timing error. We only experience this with GSM, 120ms packets to 20ms transcoding between other formats is fine. Has anyone else experienced this? I've experienced it on git-e7acd4d (our production kit) and f3c33c7 (due to become our next production kit) I'll continue to test it before submitting a Jira but I wanted to know if it was just us experiencing this. Regards, Dan
other formats also with 120 ms ptime? You do run the risk of losing a significant amount of audio if you drop even 1 120ms packet. The other end would be starved for 5 intervals. you could try rtp_timer_name=none on the 120ms leg On Fri, Mar 25, 2011 at 2:22 PM, Dan Lane <null@invalid.name> wrote:
I need to do a bit more work on this before I have enough information to submit a Jira but just on the off-chance anyone else has had this issue:
We have handsets using GSM with 120ms ptime but when this gets transcoded to something else (for example 20ms PCMU) the resulting audio is choppy as though there was a timing error. We only experience this with GSM, 120ms packets to 20ms transcoding between other formats is fine.
Has anyone else experienced this? I've experienced it on git-e7acd4d (our production kit) and f3c33c7 (due to become our next production kit)
I'll continue to test it before submitting a Jira but I wanted to know if it was just us experiencing this.
Regards, Dan
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On Fri, Mar 25, 2011 at 8:39 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
other formats also with 120 ms ptime?
Namely PCMA / PCMU. We've noticed that G722 is limited to 60ms and I don't seem to be able to get iLBC to work above 40ms.
You do run the risk of losing a significant amount of audio if you drop even 1 120ms packet. The other end would be starved for 5 intervals.
Yes, but we're dealing with mobile endpoints behind very jittery connections.
you could try rtp_timer_name=none on the 120ms leg
That fixed it... rtp_timer_name doesn't appear to be documented very well on the wiki so are there any negative side-effects to having this set to none? (I'll update the wiki with anything I learn as I'm doing this). Thanks :)
The only negative side effect is that with no timer, the channel only times off the inbound stream so if there is no rtp it would not playback sound. This is similar to how asterisk behaves by default. It's not noticeable at all during bridge but could be noticeable while on hold if you have vad enabled. On Mon, Mar 28, 2011 at 9:14 AM, Dan Lane <null@invalid.name> wrote:
On Fri, Mar 25, 2011 at 8:39 PM, Anthony Minessale <anthony.minessale@gmail.com> wrote:
other formats also with 120 ms ptime?
Namely PCMA / PCMU. We've noticed that G722 is limited to 60ms and I don't seem to be able to get iLBC to work above 40ms.
You do run the risk of losing a significant amount of audio if you drop even 1 120ms packet. The other end would be starved for 5 intervals.
Yes, but we're dealing with mobile endpoints behind very jittery connections.
you could try rtp_timer_name=none on the 120ms leg
That fixed it... rtp_timer_name doesn't appear to be documented very well on the wiki so are there any negative side-effects to having this set to none? (I'll update the wiki with anything I learn as I'm doing this).
Thanks :)
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you could try rtp_timer_name=none on the 120ms leg
That fixed it... rtp_timer_name doesn't appear to be documented very well on the wiki so are there any negative side-effects to having this set to none? (I'll update the wiki with anything I learn as I'm doing this).
Dan, Thanks for helping w/ the wiki. Let me know if you have any questions about editing on the wiki. -MC
participants (3)
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Anthony Minessale
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Dan Lane
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Michael Collins