# Asterisk RTP with WebRTC **URL:** \ **Category:** Asterisk WebRTC\ **Created:** [February 16, 2025, 10:13am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978 "2025-02-16T10:13:53Z")\ **Posts on this page:** 11\ **Page:** 1
**Author:** ![henryhkust](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/henryhkust/32/8824_2.png) [@henryhkust](https://community.asterisk.org/u/henryhkust)\ **Post date:** [February 16, 2025, 10:13am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/1 "2025-02-16T10:13:53Z")
Hi, I am building a website to make outbound calls with Asterisk as a gateway. I used JsSIP+Websocket secure for frontend WebRTC setting and configured pjsip.conf, extensions.conf, http.conf. I have three SIP accounts in pjsip.conf: a WebRTC account, a local SIP account and a SIP account from SIP2SIP.info: [webrtc\_client] type=endpoint aors=webrtc\_client auth=webrtc\_client dtls\_auto\_generate\_cert=yes webrtc=yes context=from-internal disallow=all allow=ulaw force\_rport=yes rewrite\_contact=yes transport=transport-udp [webrtc\_client] type=aor max\_contacts=5 remove\_existing=yes [webrtc\_client] type=auth auth\_type=userpass username=webrtc\_client password=\*\*\*\*\*\* ; [6002] type=endpoint context=from-internal disallow=all allow=ulaw auth=6002-auth aors=6002 direct\_media=yes transport=transport-udp [6002-auth] type=auth auth\_type=userpass username=6002 password=\*\*\*\*\*\* [6002] type=aor max\_contacts=2 [sip2sip\_provider] type = endpoint context = outbound disallow = all allow = ulaw outbound\_auth = sip2sip\_provider\_auth aors = sip2sip\_provider\_aor rtp\_symmetric=yes force\_rport=yes direct\_media=no rtcp\_mux=yes ice\_support=yes transport=transport-tcp from\_user=\*\*\*\*\*\* from\_domain=sip2sip.info [sip2sip\_provider\_auth] type = auth auth\_type = userpass username = sip2sip\_account password = \*\*\*\*\*\* [sip2sip\_provider\_aor] type = aor contact = sip:sip2sip.info [sip2sip\_provider\_registration] type = registration outbound\_auth = sip2sip\_provider\_auth server\_uri = sip:sip2sip.info client\_uri = sip:\*\*\*\*\*\*@sip2sip.info retry\_interval = 60 [sip2sip\_identify] type=identify endpoint=sip2sip\_provider match=sip2sip.info I made a call between webrtc\_client (browser) and 6002 (logged on Zoiper) with normal audio. However, when I call from webrtc\_client (via sip2sip\_account) to another SIP2SIP account on Zoiper, the SIP invitation is fine but when I accept the call, there is no media at all. I checked pjsip registrations and endpoints, with no problem. I also added NAT-related configuration in rtp.conf: [general] [stunaddr:3478=stun.voip.blackberry.com](http://stunaddr:3478=stun.voip.blackberry.com) rtpstart=10000 rtpend=20000 and pjsip.conf: [general] [stunaddr:3478=stun.voip.blackberry.com](http://stunaddr:3478=stun.voip.blackberry.com) rtpstart=10000 rtpend=20000 externip=\*\*\*\*\*\* localnet=172. **.**.\*\*/255.255.255.0 nat=force\_rport,comedia codec\_translate=yes tls\_server=yes tls\_cert\_file=/etc/asterisk/keys/asterisk.pem tls\_private\_key=/etc/asterisk/keys/asterisk.key [transport-udp] type=transport protocol=udp bind=0.0.0.0 local\_net=172. **.**.**/24 local\_net=127.0.0.1/32 external\_media\_address=**\*\*\*\* external\_signaling\_address=\*\*\*\*\*\* [transport-tcp] type=transport protocol=tcp bind=0.0.0.0 local\_net=172. **.**.**/24 external\_media\_address=**\*\*\*\* external\_signaling\_address=\*\*\*\*\*\* I used rtp set debug on and see that when I established the call to the SIP2SIP account, there is no rtp channel built. I wonder what makes my outbound call (to SIP2SIP.info) with no audio but the local call is fine? Is it because I used a self-signed certificate or the NAT configuration is not properly set? Thanks! ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [February 16, 2025, 11:26am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/2 "2025-02-16T11:26:03Z")
Please provide “pjsip set logger on” type logs. I don’t think the certificate is an issue. Certificate problems generally result in the receiving end not seeing any signalling. Also “self signed” is misused in the Asterisk world. Typically the certificates are signed by a corporate signing certificate, which might be a root certficate, and therefore, like all root certificates, sel signed, but is not th e certificate actually. Although not relevant to the issue, context=outbound is confusing. context relates to incoming requests and when used in relation to a provider, it will apply to calls that are incoming to your site. I’m not sure that chan\_pjsip has a general section, at all, but it certainly doesn’t have a parameter called nat. I don’t think WebRTC supports TLS connections being created in the to browser direction, so I don’t think rewrite\_contact serves any purpose. Given those, I’d suggest you check all the settings against the documentation, and make sure they are valid and relevant. ---
**Author:** ![henryhkust](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/henryhkust/32/8824_2.png) [@henryhkust](https://community.asterisk.org/u/henryhkust)\ **Post date:** [February 17, 2025, 2:45am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/3 "2025-02-17T02:45:44Z")
Thanks for you help! The pjsip set logger on is partly as below: ![asterisk3-imageonline.co-merged](https://global.discourse-cdn.com/asterisk/original/2X/f/f955c96dff4e627cc0ab4dc84972fb70cf563733.jpeg) I compared it with the normal internal call and found they are almost the same. However, with “rtp set debug on”, the internal call will immediately show a lot: Got RTP packet from 172.16._._:49685 (type 00, seq 037118, ts 1795590354, len 000160) Sent RTP packet to 172.16._._:46864 (via ICE) (type 00, seq 011646, ts 1795590352, len 000170) right after the two accounts joined ‘simple\_bridge’ basic-bridge. There is no rtp packet for the outbound call. For context=outbound, I mean the dial plan in the extensions.conf: [outbound] exten =\> \_[a-z].,1,Dial(PJSIP/${EXTEN}@sip2sip\_provider,30) exten =\> \_X.,n,Hangup() I will check those settings like nat, and rewrite\_contact from the documentation. Thx! ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [February 17, 2025, 8:57am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/4 "2025-02-17T08:57:02Z")
Please use plain text, not images, for logs. The log of the outbound call is incomplete, and missing an important part. However, you need to explain why the OK comes from a different address from the media address, in its SDP. ---
**Author:** ![henryhkust](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/henryhkust/32/8824_2.png) [@henryhkust](https://community.asterisk.org/u/henryhkust)\ **Post date:** [February 17, 2025, 10:15am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/5 "2025-02-17T10:15:28Z")
I tried to copy the plain text but the community does not allow more than two links published by a new account. The different address do you mean the contact: sip: another\_account@36.7._._? This address is the external IP while 172.16._._ is the internal IP behind NAT. The complete log is shown below: Executing [another\_account@from-internal:1] Dial(“PJSIP/webrtc\_client-00000004”, “PJSIP/another\_account@sip2sip\_provider,30”) in new stack – Called PJSIP/another\_account@sip2sip\_provider \<— Transmitting SIP request (1269 bytes) to TCP:212.95.45.179:5060 —\> INVITE sip:another\_account@sip2sip.info SIP/2.0 Via: SIP/2.0/TCP 36.7._._:5060;rport;branch=z9hG4bKPjf178dde7-80e0-424e-b708-c47f91e4d73d;alias From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 To: sip:another\_account@sip2sip.info Contact: sip:sip2sip\_account@36.7.:5060\*.\*;transport=TCP Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27154 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 22.1.1 Content-Type: application/sdp Content-Length: 580 v=0 o=- 420294206 420294206 IN IP4 36.7._._ s=Asterisk c=IN IP4 36.7._._ t=0 0 m=audio 13112 RTP/AVP 0 101 a=ice-ufrag:5b13cc4e08a228cc58eeab973509d39c a=ice-pwd:54ec330c1e811d4033059d0c09e46760 a=candidate:Hac100935 1 UDP 2130706431 172.16._._ 13112 typ host a=candidate:H4742a809 1 UDP 2130706431 fe80::1f08:33aa:e8f:c545 13112 typ host a=candidate:S24079f4e 1 UDP 1694498815 36.7._._ 15326 typ srflx raddr 172.16._._ rport 13112 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv a=rtcp-mux \<— Received SIP response (389 bytes) from TCP:212.95.45.179:5060 —\> SIP/2.0 100 Giving it a try Via: SIP/2.0/TCP 36.7._._:5060;received=36.7._._;rport=38534;branch=z9hG4bKPjf178dde7-80e0-424e-b708-c47f91e4d73d;alias To: sip:another\_account@sip2sip.info From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27154 INVITE Server: SIP Thor on OpenSIPS XS 3.2 Content-Length: 0 \<— Received SIP response (569 bytes) from TCP:212.95.45.179:5060 —\> SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/TCP 36.7._._:5060;received=36.7._._;rport=38534;branch=z9hG4bKPjf178dde7-80e0-424e-b708-c47f91e4d73d;alias To: sip:another\_account@sip2sip.info;tag=424d.4e7b475c2e0ef26b35bf812286b0ec69 From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27154 INVITE Proxy-Authenticate: Digest realm=“sip2sip.info”, nonce=“kR2SvyBkIbok1PlvcWcT7PTdRQ85R20YaeZVzHIYXdkA”, qop=“auth,auth-int” Server: SIP Thor on OpenSIPS XS 3.2 Content-Length: 0 \<— Transmitting SIP request (423 bytes) to TCP:212.95.45.179:5060 —\> ACK sip:another\_account@sip2sip.info SIP/2.0 Via: SIP/2.0/TCP 36.7._._:5060;rport;branch=z9hG4bKPjf178dde7-80e0-424e-b708-c47f91e4d73d;alias From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 To: sip:another\_account@sip2sip.info;tag=424d.4e7b475c2e0ef26b35bf812286b0ec69 Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27154 ACK Max-Forwards: 70 User-Agent: Asterisk PBX 22.1.1 Content-Length: 0 \<— Transmitting SIP request (1537 bytes) to TCP:212.95.45.179:5060 —\> INVITE sip:another\_account@sip2sip.info SIP/2.0 Via: SIP/2.0/TCP 36.7._._:5060;rport;branch=z9hG4bKPj6cfd673b-6745-44b1-982e-97c8efcb37d8;alias From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 To: sip:another\_account@sip2sip.info Contact: sip:sip2sip\_account@36.7.:5060\*.\*;transport=TCP Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27155 INVITE Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Supported: 100rel, timer, replaces, norefersub, histinfo Session-Expires: 1800 Min-SE: 90 Max-Forwards: 70 User-Agent: Asterisk PBX 22.1.1 Proxy-Authorization: Digest username=“sip2sip\_account”, realm=“sip2sip.info”, nonce=“kR2SvyBkIbok1PlvcWcT7PTdRQ85R20YaeZVzHIYXdkA”, uri=“sip:another\_account@sip2sip.info”, response=“b93a43667e48ea4a76171d9662f651b9”, cnonce=“7d04b044e3884276afc66d97769bd1a3”, qop=auth, nc=00000001 Content-Type: application/sdp Content-Length: 580 v=0 o=- 420294206 420294206 IN IP4 36.7._._ s=Asterisk c=IN IP4 36.7._._ t=0 0 m=audio 13112 RTP/AVP 0 101 a=ice-ufrag:5b13cc4e08a228cc58eeab973509d39c a=ice-pwd:54ec330c1e811d4033059d0c09e46760 a=candidate:Hac100935 1 UDP 2130706431 172.16._._ 13112 typ host a=candidate:H4742a809 1 UDP 2130706431 fe80::1f08:33aa:e8f:c545 13112 typ host a=candidate:S24079f4e 1 UDP 1694498815 36.7._._ 15326 typ srflx raddr 172.16._._ rport 13112 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=maxptime:140 a=sendrecv a=rtcp-mux \<— Received SIP response (389 bytes) from TCP:212.95.45.179:5060 —\> SIP/2.0 100 Giving it a try Via: SIP/2.0/TCP 36.7._._:5060;received=36.7._._;rport=38534;branch=z9hG4bKPj6cfd673b-6745-44b1-982e-97c8efcb37d8;alias To: sip:another\_account@sip2sip.info From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27155 INVITE Server: SIP Thor on OpenSIPS XS 3.2 Content-Length: 0 \<— Received SIP response (1210 bytes) from TCP:212.95.45.179:5060 —\> SIP/2.0 180 Ringing Via: SIP/2.0/TCP 36.7._._:5060;received=36.7._._;rport=38534;branch=z9hG4bKPj6cfd673b-6745-44b1-982e-97c8efcb37d8;alias Record-Route: sip:212.95.45.179;transport=tcp;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Record-Route: sip:212.95.45.179;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Record-Route: sip:174.142.205.46;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.7e99d37 Record-Route: sip:212.95.45.179;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Record-Route: sip:212.95.45.179;transport=tcp;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Contact: sip:another\_account@36.7.:27555\*.\*;transport=TCP To: sip:another\_account@sip2sip.info;tag=10c0877a From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27155 INVITE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri User-Agent: Z 5.6.6 v2.10.20.5 Allow-Events: presence, kpml, talk, as-feature-event Content-Length: 0 ``` -- PJSIP/sip2sip_provider-00000005 is ringing ``` \<— Transmitting SIP response (490 bytes) to WSS:172.16._._:49638 —\> SIP/2.0 180 Ringing Via: SIP/2.0/WSS fjoqd5gjb1p6.invalid;rport=49638;received=172.16._._;branch=z9hG4bK6143434 Call-ID: cgjj64hg09hvo64q00r0 From: sip:webrtc\_client@172.16.\*.\*;tag=r973a39flu To: sip:another\_account@172.16.\*.\*;tag=dcc20b1a-9d2d-40cb-9b2a-a228bbe05526 CSeq: 9821 INVITE Server: Asterisk PBX 22.1.1 Contact: sip:172.16.:5060\*.\* Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Content-Length: 0 \<— Received SIP response (1634 bytes) from TCP:212.95.45.179:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/TCP 36.7._._:5060;received=36.7._._;rport=38534;branch=z9hG4bKPj6cfd673b-6745-44b1-982e-97c8efcb37d8;alias Record-Route: sip:212.95.45.179;transport=tcp;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Record-Route: sip:212.95.45.179;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Record-Route: sip:174.142.205.46;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.7e99d37 Record-Route: sip:212.95.45.179;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Record-Route: sip:212.95.45.179;transport=tcp;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Require: timer Contact: sip:another\_account@36.7.:27555\*.\*;transport=TCP To: sip:another\_account@sip2sip.info;tag=10c0877a From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27155 INVITE Session-Expires: 1800;refresher=uac Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri User-Agent: Z 5.6.6 v2.10.20.5 Allow-Events: presence, kpml, talk, as-feature-event Content-Length: 343 v=0 o=Z 0 2756944404 IN IP4 172.16.9.52 s=Z c=IN IP4 67.205.116.200 t=0 0 m=audio 52800 RTP/AVP 0 106 9 8 3 101 98 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 a=sendrecv a=rtcp-mux \<— Transmitting SIP request (905 bytes) to TCP:212.95.45.179:5060 —\> ACK sip:another\_account@36.7._._:27555;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 36.7._._:5060;rport;branch=z9hG4bKPj732791c3-a31d-4b00-aac1-d32e20ebc72e;alias From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 To: sip:another\_account@sip2sip.info;tag=10c0877a Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27155 ACK Route: sip:212.95.45.179;transport=tcp;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Route: sip:212.95.45.179;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Route: sip:174.142.205.46;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.7e99d37 Route: sip:212.95.45.179;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Route: sip:212.95.45.179;transport=tcp;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Max-Forwards: 70 User-Agent: Asterisk PBX 22.1.1 Content-Length: 0 ``` -- PJSIP/sip2sip_provider-00000005 answered PJSIP/webrtc_client-00000004 ``` \<— Transmitting SIP response (1555 bytes) to WSS:172.16._._:49638 —\> SIP/2.0 200 OK Via: SIP/2.0/WSS fjoqd5gjb1p6.invalid;rport=49638;received=172.16._._;branch=z9hG4bK6143434 Call-ID: cgjj64hg09hvo64q00r0 From: sip:webrtc\_client@172.16.\*.\*;tag=r973a39flu To: sip:another\_account@172.16.\*.\*;tag=dcc20b1a-9d2d-40cb-9b2a-a228bbe05526 CSeq: 9821 INVITE Server: Asterisk PBX 22.1.1 Allow: OPTIONS, REGISTER, SUBSCRIBE, NOTIFY, PUBLISH, INVITE, ACK, BYE, CANCEL, UPDATE, PRACK, INFO, MESSAGE, REFER Contact: sip:172.16.:5060\*.\* Supported: 100rel, timer, replaces, norefersub Session-Expires: 90;refresher=uac Require: timer Content-Type: application/sdp Content-Length: 937 v=0 o=- 3064053736346234544 4 IN IP4 172.16._._ s=Asterisk c=IN IP4 172.16._._ t=0 0 a=msid-semantic:WMS \* a=group:BUNDLE 0 m=audio 18496 UDP/TLS/RTP/SAVPF 0 126 a=connection:new a=setup:active a=fingerprint:SHA-256 47:D2:2F:CB:44:63:C4:AE:BA:BE:B4:A2:48:1F:22:01:7A:D7:6E:D6:E0:DA:EA:2F:0F:43:85:58:E2:84:69:24 a=ice-ufrag:2e5d32be54af2edb7f3b934f55a8a1a1 a=ice-pwd:575112a551e8f37b459966d25b989bbf a=candidate:Hac100935 1 UDP 2130706431 172.16._._ 18496 typ host a=candidate:S24079f4e 1 UDP 1694498815 36.7._._ 15315 typ srflx raddr 172.16._._ rport 18496 a=rtpmap:0 PCMU/8000 a=rtpmap:126 telephone-event/8000 a=fmtp:126 0-16 a=ptime:20 a=maxptime:140 a=sendrecv a=rtcp-mux a=ssrc:362561819 cname:d0c4184f-806a-47c0-8341-1ba343898657 a=msid:6490807e-13a8-4c45-ab47-45fb3d599963 6d66a681-73a8-4176-ab1b-dec135ade4e6 a=rtcp-fb:\* transport-cc a=mid:0 m=audio 0 UDP/TLS/RTP/SAVPF 111 63 9 0 8 13 110 126 ``` -- Channel PJSIP/sip2sip_provider-00000005 joined 'simple_bridge' basic-bridge <939c157d-45a0-40ba-9ba9-e2c3b93e6f8a> -- Channel PJSIP/webrtc_client-00000004 joined 'simple_bridge' basic-bridge <939c157d-45a0-40ba-9ba9-e2c3b93e6f8a> ``` \<— Received SIP request (415 bytes) from WSS:172.16._._:49638 —\> ACK sip:172.16._._:5060 SIP/2.0 Via: SIP/2.0/WSS fjoqd5gjb1p6.invalid;branch=z9hG4bK4796425 Max-Forwards: 69 To: sip:another\_account@172.16.\*.\*;tag=dcc20b1a-9d2d-40cb-9b2a-a228bbe05526 From: sip:webrtc\_client@172.16.\*.\*;tag=r973a39flu Call-ID: cgjj64hg09hvo64q00r0 CSeq: 9821 ACK Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: outbound User-Agent: JsSIP 3.1.1 Content-Length: 0 \<— Received SIP response (1634 bytes) from TCP:212.95.45.179:5060 —\> SIP/2.0 200 OK Via: SIP/2.0/TCP 36.7._._:5060;received=36.7._._;rport=38534;branch=z9hG4bKPj6cfd673b-6745-44b1-982e-97c8efcb37d8;alias Record-Route: sip:212.95.45.179;transport=tcp;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Record-Route: sip:212.95.45.179;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Record-Route: sip:174.142.205.46;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.7e99d37 Record-Route: sip:212.95.45.179;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Record-Route: sip:212.95.45.179;transport=tcp;r2=on;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Require: timer Contact: sip:another\_account@36.7.:27555\*.\*;transport=TCP To: sip:another\_account@sip2sip.info;tag=10c0877a From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27155 INVITE Session-Expires: 1800;refresher=uac Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, MESSAGE, OPTIONS, INFO, SUBSCRIBE Content-Type: application/sdp Supported: replaces, norefersub, extended-refer, timer, sec-agree, outbound, path, X-cisco-serviceuri User-Agent: Z 5.6.6 v2.10.20.5 Allow-Events: presence, kpml, talk, as-feature-event Content-Length: 343 v=0 o=Z 0 2756944404 IN IP4 172.16.9.52 s=Z c=IN IP4 67.205.116.200 t=0 0 m=audio 52800 RTP/AVP 0 106 9 8 3 101 98 a=rtpmap:106 opus/48000/2 a=fmtp:106 sprop-maxcapturerate=16000; minptime=20; useinbandfec=1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:98 telephone-event/48000 a=fmtp:98 0-16 a=sendrecv a=rtcp-mux \<— Transmitting SIP request (905 bytes) to TCP:212.95.45.179:5060 —\> ACK sip:another\_account@36.7._._:27555;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 36.7._._:5060;rport;branch=z9hG4bKPj732791c3-a31d-4b00-aac1-d32e20ebc72e;alias From: sip:sip2sip\_account@sip2sip.info;tag=add5ef57-107a-40ee-ac41-c776e1de09c6 To: sip:another\_account@sip2sip.info;tag=10c0877a Call-ID: cef2e79c-2334-416b-9e2a-16b73b80b4cf CSeq: 27155 ACK Route: sip:212.95.45.179;transport=tcp;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Route: sip:212.95.45.179;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.56b32013 Route: sip:174.142.205.46;lr;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.7e99d37 Route: sip:212.95.45.179;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Route: sip:212.95.45.179;transport=tcp;lr;r2=on;ftag=add5ef57-107a-40ee-ac41-c776e1de09c6;did=e.66b32013 Max-Forwards: 70 User-Agent: Asterisk PBX 22.1.1 Content-Length: 0 \<— Received SIP request (413 bytes) from WSS:172.16._._:49638 —\> BYE sip:172.16._._:5060 SIP/2.0 Via: SIP/2.0/WSS fjoqd5gjb1p6.invalid;branch=z9hG4bK85992 Max-Forwards: 69 To: sip:another\_account@172.16.\*.\*;tag=dcc20b1a-9d2d-40cb-9b2a-a228bbe05526 From: sip:webrtc\_client@172.16.\*.\*;tag=r973a39flu Call-ID: cgjj64hg09hvo64q00r0 CSeq: 9822 BYE Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS,REFER,INFO Supported: outbound User-Agent: JsSIP 3.1.1 Content-Length: 0 ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [February 17, 2025, 2:16pm UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/6 "2025-02-17T14:16:17Z")
> [@henryhkust](#): > > The different address do you mean the contact: sip: another\_account@36.7._._? This address is the external IP while 172.16._._ is the internal IP behind NAT. I mean > [@henryhkust](#): > > from TCP:212.95 doesn’t match > [@henryhkust](#): > > c=IN IP4 67.205. This is a broken NAT configuration at the remote side, which means that, for media to work, both Asterisk has to be configured for symmetric\_rtp, and the they have to send media to Asterisk first, so that Asterisk can learn their public media address. ---
**Author:** ![henryhkust](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/henryhkust/32/8824_2.png) [@henryhkust](https://community.asterisk.org/u/henryhkust)\ **Post date:** [February 18, 2025, 10:34am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/7 "2025-02-18T10:34:22Z")
Thanks for your clarification. I checked the two IP addresses: 212.95.45.179 and 67.205.116.200. Neither of them belongs to my part. I think they are the servers from sip2sip or Zoiper. I wonder what should be the IP in “c=IN IP4 67.205.116.200”, my external IP or the IP of the other side? Do you mean the problem is from the other side (sip2sip)? My intended workflow is: dial a number from the website (webrtc\_account) =\> use the registered sip2sip account (sip2sip\_account) to make the call =\> talk with another sip2sip account (another\_account). ---
**Author:** ![henryhkust](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/henryhkust/32/8824_2.png) [@henryhkust](https://community.asterisk.org/u/henryhkust)\ **Post date:** [February 18, 2025, 11:49am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/8 "2025-02-18T11:49:33Z")
I solved the problem. I replaced Zoiper with Linephone and the RTP is fine now. Very strange! ---
**Author:** ![david551](https://avatars.discourse-cdn.com/v4/letter/d/3bc359/32.png) [@david551](https://community.asterisk.org/u/david551)\ **Post date:** [February 18, 2025, 12:27pm UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/9 "2025-02-18T12:27:29Z")
> [@henryhkust](#): > > I wonder what should be the IP in “c=IN IP4 67.205.116.200”, my external IP or the IP of the other side? In that particular one, the external IP of the other side, as it is the SDP coming from them. ---
**Author:** ![henryhkust](https://sea1.discourse-cdn.com/asterisk/user_avatar/community.asterisk.org/henryhkust/32/8824_2.png) [@henryhkust](https://community.asterisk.org/u/henryhkust)\ **Post date:** [February 19, 2025, 5:38am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/10 "2025-02-19T05:38:54Z")
Got it. So the problem is indeed caused by the server on the Zoiper side, right? Anyway, thank you for your kind help, David! ---
**Author:** ![system](https://global.discourse-cdn.com/asterisk/original/2X/4/475d0c889911cfb84ac4881ad6df8b523e3f9105.png) [@system](https://community.asterisk.org/u/system)\ **Post date:** [March 21, 2025, 5:39am UTC](https://community.asterisk.org/t/asterisk-rtp-with-webrtc/106978/11 "2025-03-21T05:39:33Z")
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