# Opus codec bandwidth **URL:** \ **Category:** Asterisk Support\ **Created:** [February 4, 2019, 12:35pm UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388 "2019-02-04T12:35:59Z")\ **Posts on this page:** 9\ **Page:** 1
**Author:** ![Rozwielitka](https://avatars.discourse-cdn.com/v4/letter/r/a8b319/32.png) [@Rozwielitka](https://community.asterisk.org/u/Rozwielitka)\ **Post date:** [February 4, 2019, 12:35pm UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/1 "2019-02-04T12:35:59Z")
Hi, Im not sure why, but Opus codec is ignoring max\_bandwidth parameter. Parameter can be added in codecs.conf, but if you will check tcpdump trace, this parameter is ignored. In codecs.conf im adding: [opus8] type=opus bitrate=8000 cbr=yes packet\_loss=5 max\_bandwidth=narrow endpoints for sure have this codec assigned using: allow=opus8 But if you will start tcpdump and capture such call you will find that used bandwidth is around 60kbps in case of SRTP (in wireshark statistics-\>protocol hierarchy). 60kbps is very far from 8kbps, Is there any way to limit bandwidth for opus? Please help, Best regards and thanks, ---
**Author:** ![kharwell](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kharwell/32/1924_2.png) [@kharwell](https://community.asterisk.org/u/kharwell)\ **Post date:** [February 4, 2019, 5:20pm UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/2 "2019-02-04T17:20:29Z")
Please post the sip trace of the call. Also, were the observed packets coming from or going to Asterisk? ---
**Author:** ![Rozwielitka](https://avatars.discourse-cdn.com/v4/letter/r/a8b319/32.png) [@Rozwielitka](https://community.asterisk.org/u/Rozwielitka)\ **Post date:** [February 5, 2019, 8:03am UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/3 "2019-02-05T08:03:40Z")
Hello, Please find enclosed sip and rtp trace (because of sire limitation i wasnt able to enclose capture). [res\_txt.txt](https://global.discourse-cdn.com/asterisk/original/2X/0/0e5e867bea647fe76ee631a1549bcb6cc19ff7d9.txt) (917.2 KB) In INVITE and Ok200 you will see that opus was negociated, Checking Statistics-\>Protocol Hierarchy is showing that RTP for this call (in both directions) is using 43kbps. If you will replace rtp with srtp single call will use 67kbps-90kbps. so the limit: bitrate=8000 max\_bandwidth=narrow Does not work. In 4G environment it will work, but for most 2G and some 2.5G/3G it wont. Is ther any way to limit opus traffic per call? I check this for both chan sip and chan pjsip. Behaviour is the same. Best regards, ---
**Author:** ![Rozwielitka](https://avatars.discourse-cdn.com/v4/letter/r/a8b319/32.png) [@Rozwielitka](https://community.asterisk.org/u/Rozwielitka)\ **Post date:** [February 5, 2019, 8:57am UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/4 "2019-02-05T08:57:39Z")
Codec was installed from [downloads.digium.com/pub/telephony/codec\_opus:](http://downloads.digium.com/pub/telephony/codec_opus:) wget [http://downloads.digium.com/pub/telephony/codec\_opus/asterisk-13.0/x86-64/codec\_opus-13.0\_current-x86\_64.tar.gz](http://downloads.digium.com/pub/telephony/codec_opus/asterisk-13.0/x86-64/codec_opus-13.0_current-x86_64.tar.gz) tar -xvzf codec\_opus-13.0\_current-x86\_64.tar.gz cd codec\_opus-13.0\_1.1.0-x86\_64 cp codec\_opus.so /usr/lib/asterisk/modules/ cp codec\_opus\_config-en\_US.xml /var/lib/asterisk/documentation/thirdparty This behaviour is reproducable for both asterisk 13 and 15. ---
**Author:** ![kharwell](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kharwell/32/1924_2.png) [@kharwell](https://community.asterisk.org/u/kharwell)\ **Post date:** [February 5, 2019, 4:35pm UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/5 "2019-02-05T16:35:26Z")
Unfortunately that trace does not have the SDP included. Can you post the SDP for the invite and 200 OK. If it’s easier you can get a sip trace directly from the Asterisk log/console by turning on sip debugging for the channel driver. For instance: ``` *CLI> pjsip set logger on ``` ---
**Author:** ![Rozwielitka](https://avatars.discourse-cdn.com/v4/letter/r/a8b319/32.png) [@Rozwielitka](https://community.asterisk.org/u/Rozwielitka)\ **Post date:** [February 6, 2019, 6:06am UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/6 "2019-02-06T06:06:20Z")
Sorry for delay, Please find enclosed SIP trace with SDP. This is incoming call, so SDP from asterisk is in OK 200: \<— SIP read from UDP:192.168.1.173:48866 —\> INVITE [sip:999@192.168.13.33](mailto:sip:999@192.168.13.33) SIP/2.0 Via: SIP/2.0/UDP 10.0.2.15:1043;rport;branch=z9hG4bKPj483cbb9e2ff64ac295e4c88f9ed3be8b Max-Forwards: 70 From: sip:504@192.168.13.33;tag=65def2f85f21457e8ce6a80d015ac579 To: sip:999@192.168.13.33 Contact: sip:504@10.0.2.15:1043;ob Call-ID: 283a025397fe426d94298469aabd9503 CSeq: 5536 INVITE Allow: PRACK, INVITE, ACK, BYE, CANCEL, UPDATE, INFO, SUBSCRIBE, NOTIFY, REFER, MESSAGE, OPTIONS Supported: replaces, 100rel, timer, norefersub Session-Expires: 1800 Min-SE: 90 User-Agent: MicroSIP/3.19.10 Authorization: Digest username=“504”, realm=“asterisk”, nonce=“534f3196”, [uri="sip:999@192.168.13.33](mailto:uri=%22sip:999@192.168.13.33)", response=“3817f473a850db5f180372b4fd3cab2d”, algorithm=MD5 Content-Type: application/sdp Content-Length: 414 v=0 o=- 3758425327 3758425327 IN IP4 10.0.2.15 s=pjmedia b=AS:67 t=0 0 a=X-nat:0 m=audio 4002 RTP/AVP 123 101 c=IN IP4 10.0.2.15 b=TIAS:48000 a=rtcp:4003 IN IP4 10.0.2.15 a=sendrecv a=rtpmap:123 opus/48000/2 a=fmtp:123 maxplaybackrate=24000;sprop-maxcapturerate=24000;maxaveragebitrate=20000;useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ssrc:1741099167 cname:1fd556cd0feb13f5 \<-------------\> — (16 headers 16 lines) — Sending to 192.168.1.173:48866 (NAT) Using INVITE request as basis request - 283a025397fe426d94298469aabd9503 Found peer ‘504’ for ‘504’ from 192.168.1.173:48866 == Using SIP RTP CoS mark 5 Found RTP audio format 123 Found RTP audio format 101 Found audio description format opus for ID 123 Found audio description format telephone-event for ID 101 Capabilities: us - (opus8|ulaw|alaw|g722|gsm|speex|ilbc), peer - audio=(opus)/video=(nothing)/text=(nothing), combined - (opus8) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 10.0.2.15:4002 Looking for 999 in mcs-ctx (domain 192.168.13.33) sip\_route\_dump: route/path hop: sip:504@10.0.2.15:1043;ob \<— Transmitting (NAT) to 192.168.1.173:48866 —\> SIP/2.0 100 Trying Via: SIP/2.0/UDP 10.0.2.15:1043;branch=z9hG4bKPj483cbb9e2ff64ac295e4c88f9ed3be8b;received=192.168.1.173;rport=48866 From: sip:504@192.168.13.33;tag=65def2f85f21457e8ce6a80d015ac579 To: sip:999@192.168.13.33 Call-ID: 283a025397fe426d94298469aabd9503 CSeq: 5536 INVITE Server: Asterisk PBX 13.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:999@192.168.13.33:5060 Content-Length: 0 \<------------\> – Executing [999@mcs-ctx:1] Answer(“SIP/504-00000000”, “”) in new stack Audio is at 19174 Adding codec opus8 to SDP Adding codec ulaw to SDP Adding codec alaw to SDP Adding codec g722 to SDP Adding codec gsm to SDP Adding codec speex to SDP Adding codec ilbc to SDP Adding non-codec 0x1 (telephone-event) to SDP \<— Reliably Transmitting (NAT) to 192.168.1.173:48866 —\> SIP/2.0 200 OK Via: SIP/2.0/UDP 10.0.2.15:1043;branch=z9hG4bKPj483cbb9e2ff64ac295e4c88f9ed3be8b;received=192.168.1.173;rport=48866 From: sip:504@192.168.13.33;tag=65def2f85f21457e8ce6a80d015ac579 To: sip:999@192.168.13.33;tag=as43be82d6 Call-ID: 283a025397fe426d94298469aabd9503 CSeq: 5536 INVITE Server: Asterisk PBX 13.6.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Session-Expires: 1800;refresher=uas Contact: sip:999@192.168.13.33:5060 Content-Type: application/sdp Require: timer Content-Length: 560 v=0 o=root 1372000500 1372000500 IN IP4 192.168.13.33 s=Asterisk PBX 13.6.0 c=IN IP4 192.168.13.33 t=0 0 m=audio 19174 RTP/AVP 123 0 8 9 3 110 97 101 a=rtpmap:123 opus/48000/2 a=fmtp:123 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:97 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:20 a=sendrecv \<------------\> \<— SIP read from UDP:192.168.1.173:48866 —\> ACK sip:999@192.168.13.33:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.15:1043;rport;branch=z9hG4bKPjb3a6f84b4173499f82020b6a65093b02 Max-Forwards: 70 From: sip:504@192.168.13.33;tag=65def2f85f21457e8ce6a80d015ac579 To: sip:999@192.168.13.33;tag=as43be82d6 Call-ID: 283a025397fe426d94298469aabd9503 CSeq: 5536 ACK Content-Length: 0 \<-------------\> — (8 headers 0 lines) — \> 0x7f4c28009e50 – Probation passed - setting RTP source address to 192.168.1.173:58663 – Executing [999@mcs-ctx:2] Echo(“SIP/504-00000000”, “”) in new stack \> 0x7f4c28009e50 – Probation passed - setting RTP source address to 192.168.1.173:58663 [Feb 6 07:37:43] WARNING[19517]: chan\_mgcp.c:667 retrans\_pkt: Maximum retries exceeded for transaction 4 on [192.168.1.137] [Feb 6 07:37:43] NOTICE[19517]: chan\_mgcp.c:2834 handle\_response: Transaction 4 timed out [Feb 6 07:37:44] WARNING[19517]: chan\_mgcp.c:667 retrans\_pkt: Maximum retries exceeded for transaction 5 on [192.168.1.137] [Feb 6 07:37:44] NOTICE[19517]: chan\_mgcp.c:2834 handle\_response: Transaction 5 timed out \<— SIP read from UDP:192.168.1.173:48866 —\> BYE sip:999@192.168.13.33:5060 SIP/2.0 Via: SIP/2.0/UDP 10.0.2.15:1043;rport;branch=z9hG4bKPj0bc8886caad040daaa76ac7ca3054e6a Max-Forwards: 70 From: sip:504@192.168.13.33;tag=65def2f85f21457e8ce6a80d015ac579 To: sip:999@192.168.13.33;tag=as43be82d6 Call-ID: 283a025397fe426d94298469aabd9503 CSeq: 5537 BYE User-Agent: MicroSIP/3.19.10 Content-Length: 0 Used config in codecs.conf: [opus8] type=opus bitrate=8000 cbr=yes packet\_loss=5 max\_bandwidth=narrow max\_playback\_rate=8000 For this test i used chan\_sip. But in chan pjsip (on my diffrent installation) behaviour is the same. max\_playback\_rate is ignored. In config is 8000, but in SDP is: a=fmtp:123 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 This endpoint config: my-codecs ; a template for my preferred codecs disallow=all allow=opus8 allow=ulaw allow=alaw allow=g722 allow=gsm allow=speex allow=ilbc allow=isac natted-phone ; another template inheriting basic-options directmedia=no host=dynamic nat=force\_rport,comedia 504 secret = pass mailbox=504 callerid=“Test1 \<504\>” Best regards ---
**Author:** ![kharwell](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kharwell/32/1924_2.png) [@kharwell](https://community.asterisk.org/u/kharwell)\ **Post date:** [February 6, 2019, 7:05pm UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/7 "2019-02-06T19:05:49Z")
So Asterisk’s response is odd: > [@Rozwielitka](#): > > a=rtpmap:123 opus/48000/2 > a=fmtp:123 maxplaybackrate=48000;sprop-maxcapturerate=48000;minptime=10;maxaveragebitrate=20000;stereo=0;sprop-stereo=0;cbr=0;useinbandfec=0;usedtx=0 Several of the fmtp parameters do not match with your configuration. Further Asterisk does not add default parameters to the fmtp line. Nor is minptime even added to the line ever. What version of Asterisk are you running? Note, the reason you are seeing your rates is because according to the SDP your maxplaybackrate is 48,000 thus the codec will operate up to full band ---
**Author:** ![Rozwielitka](https://avatars.discourse-cdn.com/v4/letter/r/a8b319/32.png) [@Rozwielitka](https://community.asterisk.org/u/Rozwielitka)\ **Post date:** [February 7, 2019, 5:52am UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/8 "2019-02-07T05:52:34Z")
This test was made on Asterisk 13.6.0 and chan\_sip, But i have another installation with Asterisk 15.2.2 with chan\_pjsip and the behaviour is the same. Parameter names in codecs.conf for sure are ok, because if you will insert wrong name during service restart you will get message “unrecognized codec opus8” and all endpoints will get “unauthorized”. For both installations im using codec\_opus-13.0\_1.3.0-x86\_64. I am able to enable fec, cbr. But im not able to manage bitrate/bandwidth. Good info is that you can limit it on endpoint side and asterisk will comply. If you will set default opus on asterisk (without any customization) and you will limit bandwidth on softphone (by setting proper parameters in sdp on clients side) asterisk will use lower bandwidth. ---
**Author:** ![kharwell](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/kharwell/32/1924_2.png) [@kharwell](https://community.asterisk.org/u/kharwell)\ **Post date:** [February 7, 2019, 4:51pm UTC](https://community.asterisk.org/t/opus-codec-bandwidth/78388/9 "2019-02-07T16:51:56Z")
codec\_opus transcoding was not officially supported in Asterisk until a much later version of 13 than you are running. Also there have been several patches to the codec opus format attribute handling code since then as well. I’d recommend upgrading to the latest version of Asterisk, 13.24.1 or 16.1.1, and re-testing.