Polycom G722 negotiation
Hello all, I have several Polycom IP-650s. On the external profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT PCMU,G729 On the internal profile I have CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM The Polycoms have a similar codec priority set: v=0 o=- 1167611879 1167611879 IN IP4 10.59.1.243 s=Polycom IP Phone c=IN IP4 10.59.1.243 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000 The issue I have is that if I leave the internal profile to generous and a user places a call from the PSTN on hold which is PCMU, when the user tries to pick up the call, the Polycoms' preference then changes the call to G722 and there is no audio. If I set the profile to greedy, since the call is already PCMU Freeswitch keeps it at PCMU and audio is fine. I realize this is the intended behavior for greedy vs generous but my question is why is there no audio when switching the codecs? I didn't want to clutter this up any more but I'm happy to provide traces however the only real difference is the codec order. Thanks, Spencer -- Spencer Thomason 5Nine Solutions LLC e. sales@5ninesolutions.com p. +1.888.271.7959 f. +1.310.510.6980
What version of FreeSWITCH are you running? /b On Feb 24, 2011, at 11:45 PM, Spencer Thomason wrote:
Hello all, I have several Polycom IP-650s. On the external profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT PCMU,G729
On the internal profile I have CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM
The Polycoms have a similar codec priority set: v=0 o=- 1167611879 1167611879 IN IP4 10.59.1.243 s=Polycom IP Phone c=IN IP4 10.59.1.243 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000
The issue I have is that if I leave the internal profile to generous and a user places a call from the PSTN on hold which is PCMU, when the user tries to pick up the call, the Polycoms' preference then changes the call to G722 and there is no audio. If I set the profile to greedy, since the call is already PCMU Freeswitch keeps it at PCMU and audio is fine. I realize this is the intended behavior for greedy vs generous but my question is why is there no audio when switching the codecs? I didn't want to clutter this up any more but I'm happy to provide traces however the only real difference is the codec order.
Thanks, Spencer
freeswitch@internal> version FreeSWITCH Version 1.0.head (hacked-20110223T050150Z) On Feb 25, 2011, at 4:18 PM, Brian West wrote:
What version of FreeSWITCH are you running?
/b
On Feb 24, 2011, at 11:45 PM, Spencer Thomason wrote:
Hello all, I have several Polycom IP-650s. On the external profile I have: CODECS IN G722,PCMU,G729,GSM CODECS OUT PCMU,G729
On the internal profile I have CODECS IN G722,PCMU,G729,GSM CODECS OUT G722,PCMU,G729,GSM
The Polycoms have a similar codec priority set: v=0 o=- 1167611879 1167611879 IN IP4 10.59.1.243 s=Polycom IP Phone c=IN IP4 10.59.1.243 t=0 0 a=sendrecv m=audio 2236 RTP/AVP 9 0 18 101 a=rtpmap:9 G722/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=yes a=rtpmap:101 telephone-event/8000
The issue I have is that if I leave the internal profile to generous and a user places a call from the PSTN on hold which is PCMU, when the user tries to pick up the call, the Polycoms' preference then changes the call to G722 and there is no audio. If I set the profile to greedy, since the call is already PCMU Freeswitch keeps it at PCMU and audio is fine. I realize this is the intended behavior for greedy vs generous but my question is why is there no audio when switching the codecs? I didn't want to clutter this up any more but I'm happy to provide traces however the only real difference is the codec order.
Thanks, Spencer
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What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get clipped audio for example when G722 30ms is offered on Snoms, but they work perfectly when I force them to 20ms. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp606... Sent from the freeswitch-users mailing list archive at Nabble.com.
The devices are using 20ms G722. The strange thing is that if the call starts out as G722, everything works totally fine. The problem arises when the phone has G722 first in its prefered codec, the FS profile is set to generous, and a call starts out as PCMU. The audio is fine until someone places a call on hold. After taking a call off hold, when the FS profile is set to generous, FS honors the Polycoms preference for G722 and then tries to switch to G722. At this point there is no audio. If the sophia profile is set to greedy, as expected Freeswitch does not allow the switch to G722 and there are no audio problems. This only happens when a call starts out in something other than G722 and then tries to switch. I.e. client ---G722---> FS ---PCMU---> Provider works and client ---G722---> FS ---G722---> client works but provider ---PCMU---> FS ---PCMU---> client and then a call is placed on hold and resumed a G722 does not. Spencer On Feb 28, 2011, at 3:34 PM, peely wrote:
What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get clipped audio for example when G722 30ms is offered on Snoms, but they work perfectly when I force them to 20ms.
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp606... Sent from the freeswitch-users mailing list archive at Nabble.com.
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Are you doing this with bypass_media=true? On Mon, Feb 28, 2011 at 9:55 AM, Spencer Thomason <spencer@5ninesolutions.com> wrote:
The devices are using 20ms G722. The strange thing is that if the call starts out as G722, everything works totally fine. The problem arises when the phone has G722 first in its prefered codec, the FS profile is set to generous, and a call starts out as PCMU. The audio is fine until someone places a call on hold. After taking a call off hold, when the FS profile is set to generous, FS honors the Polycoms preference for G722 and then tries to switch to G722. At this point there is no audio. If the sophia profile is set to greedy, as expected Freeswitch does not allow the switch to G722 and there are no audio problems. This only happens when a call starts out in something other than G722 and then tries to switch. I.e. client ---G722---> FS ---PCMU---> Provider works and client ---G722---> FS ---G722---> client works but provider ---PCMU---> FS ---PCMU---> client and then a call is placed on hold and resumed a G722 does not.
Spencer
On Feb 28, 2011, at 3:34 PM, peely wrote:
What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get clipped audio for example when G722 30ms is offered on Snoms, but they work perfectly when I force them to 20ms.
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp606... Sent from the freeswitch-users mailing list archive at Nabble.com.
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No, Freeswitch is processing the media, proxy media is not on either. On Feb 28, 2011, at 7:15 PM, Anthony Minessale wrote:
Are you doing this with bypass_media=true?
On Mon, Feb 28, 2011 at 9:55 AM, Spencer Thomason <spencer@5ninesolutions.com> wrote:
The devices are using 20ms G722. The strange thing is that if the call starts out as G722, everything works totally fine. The problem arises when the phone has G722 first in its prefered codec, the FS profile is set to generous, and a call starts out as PCMU. The audio is fine until someone places a call on hold. After taking a call off hold, when the FS profile is set to generous, FS honors the Polycoms preference for G722 and then tries to switch to G722. At this point there is no audio. If the sophia profile is set to greedy, as expected Freeswitch does not allow the switch to G722 and there are no audio problems. This only happens when a call starts out in something other than G722 and then tries to switch. I.e. client ---G722---> FS ---PCMU---> Provider works and client ---G722---> FS ---G722---> client works but provider ---PCMU---> FS ---PCMU---> client and then a call is placed on hold and resumed a G722 does not.
Spencer
On Feb 28, 2011, at 3:34 PM, peely wrote:
What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get clipped audio for example when G722 30ms is offered on Snoms, but they work perfectly when I force them to 20ms.
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp606... Sent from the freeswitch-users mailing list archive at Nabble.com.
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-- Anthony Minessale II
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It seems like a bug in the polycom, hold/unhold should maintain the original SDP already negotiated. I would try updating the firmware. I suspect polycom is not really offering G.722 but is saying they want to in the sdp. Then its ignoring the G.722 because it's still expecting ULAW. We might have to make a param that does not allow codec re-negotiation once a call is up as a hack to deal with this. On Mon, Feb 28, 2011 at 1:22 PM, Spencer Thomason <spencer@5ninesolutions.com> wrote:
No, Freeswitch is processing the media, proxy media is not on either.
On Feb 28, 2011, at 7:15 PM, Anthony Minessale wrote:
Are you doing this with bypass_media=true?
On Mon, Feb 28, 2011 at 9:55 AM, Spencer Thomason <spencer@5ninesolutions.com> wrote:
The devices are using 20ms G722. The strange thing is that if the call starts out as G722, everything works totally fine. The problem arises when the phone has G722 first in its prefered codec, the FS profile is set to generous, and a call starts out as PCMU. The audio is fine until someone places a call on hold. After taking a call off hold, when the FS profile is set to generous, FS honors the Polycoms preference for G722 and then tries to switch to G722. At this point there is no audio. If the sophia profile is set to greedy, as expected Freeswitch does not allow the switch to G722 and there are no audio problems. This only happens when a call starts out in something other than G722 and then tries to switch. I.e. client ---G722---> FS ---PCMU---> Provider works and client ---G722---> FS ---G722---> client works but provider ---PCMU---> FS ---PCMU---> client and then a call is placed on hold and resumed a G722 does not.
Spencer
On Feb 28, 2011, at 3:34 PM, peely wrote:
What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get clipped audio for example when G722 30ms is offered on Snoms, but they work perfectly when I force them to 20ms.
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp606... Sent from the freeswitch-users mailing list archive at Nabble.com.
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Yes, that what I thought. I'm running the latest firmware, 3.3.1. I'll report it to them as its clearly offering a different SDP after the unhold. Yes it might be nice if Freeswitch refused to switch codec on a call that is up even if the negotiation is set to generous. Thanks for your help, Spencer On Feb 28, 2011, at 9:19 PM, Anthony Minessale wrote:
It seems like a bug in the polycom, hold/unhold should maintain the original SDP already negotiated. I would try updating the firmware. I suspect polycom is not really offering G.722 but is saying they want to in the sdp. Then its ignoring the G.722 because it's still expecting ULAW.
We might have to make a param that does not allow codec re-negotiation once a call is up as a hack to deal with this.
On Mon, Feb 28, 2011 at 1:22 PM, Spencer Thomason <spencer@5ninesolutions.com> wrote:
No, Freeswitch is processing the media, proxy media is not on either.
On Feb 28, 2011, at 7:15 PM, Anthony Minessale wrote:
Are you doing this with bypass_media=true?
On Mon, Feb 28, 2011 at 9:55 AM, Spencer Thomason <spencer@5ninesolutions.com> wrote:
The devices are using 20ms G722. The strange thing is that if the call starts out as G722, everything works totally fine. The problem arises when the phone has G722 first in its prefered codec, the FS profile is set to generous, and a call starts out as PCMU. The audio is fine until someone places a call on hold. After taking a call off hold, when the FS profile is set to generous, FS honors the Polycoms preference for G722 and then tries to switch to G722. At this point there is no audio. If the sophia profile is set to greedy, as expected Freeswitch does not allow the switch to G722 and there are no audio problems. This only happens when a call starts out in something other than G722 and then tries to switch. I.e. client ---G722---> FS ---PCMU---> Provider works and client ---G722---> FS ---G722---> client works but provider ---PCMU---> FS ---PCMU---> client and then a call is placed on hold and resumed a G722 does not.
Spencer
On Feb 28, 2011, at 3:34 PM, peely wrote:
What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get clipped audio for example when G722 30ms is offered on Snoms, but they work perfectly when I force them to 20ms.
-- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Polycom-G722-negotiation-tp606... Sent from the freeswitch-users mailing list archive at Nabble.com.
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-- Anthony Minessale II
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I'm running 3.3.1 and I'm not seeing this behavior... did you happen to not install the default sip.cfg that came with 3.3.1? /b On Feb 28, 2011, at 3:37 PM, Spencer Thomason wrote:
Yes, that what I thought. I'm running the latest firmware, 3.3.1. I'll report it to them as its clearly offering a different SDP after the unhold. Yes it might be nice if Freeswitch refused to switch codec on a call that is up even if the negotiation is set to generous.
Thanks for your help, Spencer
If you'd like I can pastebin my config files. But with 3.3.1 its my understanding that there is no sip.cfg file anymore since Polycom changed the config format. I did however base my config files off the ones in the /configs directory in the firmware zip file and specifically everything codec related has not been changed from the defaults. On Mar 1, 2011, at 12:28 PM, Brian West wrote:
I'm running 3.3.1 and I'm not seeing this behavior... did you happen to not install the default sip.cfg that came with 3.3.1?
/b
On Feb 28, 2011, at 3:37 PM, Spencer Thomason wrote:
Yes, that what I thought. I'm running the latest firmware, 3.3.1. I'll report it to them as its clearly offering a different SDP after the unhold. Yes it might be nice if Freeswitch refused to switch codec on a call that is up even if the negotiation is set to generous.
Thanks for your help, Spencer
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BZZZZT WRONG. We support multiples of 20 from 20 to 120 for most codecs. /b On Feb 28, 2011, at 9:34 AM, peely wrote:
What ptime is your device offering? FreeSWITCH only supports 20ms G722, I get clipped audio for example when G722 30ms is offered on Snoms, but they work perfectly when I force them to 20ms.
participants (4)
-
Anthony Minessale
-
Brian West
-
peely
-
Spencer Thomason